Re: [Asterisk-Users] SIPP load testing - unexpected message - anyone using sipp sucessfully ?
SIP will get you no RTP, meaning it only works with SIP headers. Asterisks CPU usage is mainly coming from RTP handling. We glued something together that will work for RTP too, you can download it from: http://www.astertest.com/forum/viewtopic.php?t=4 As the moment it only seems to work for non authenticated SIP calls, but it does support RTP. Other options are commercial tools such as WINSIP etc. (more call generators + descriptions can be found in the ppt presentation on www.astertest.com) SIPP works for asterisk testing too, but you need the correct commandline. What did you use ? Joachim Robert Rozman wrote: Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get Unexpected message for Call-ID ..., so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test Asterisk ? Is sipp the right tool ? Thanks in advance, regards, Rob. sipp: The following events occured: 2005-02-08 00:23:36: Unexpected message for Call-ID '[EMAIL PROTECTED]': while expecting '100' response, received 'SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060 From: sipp sip:[EMAIL PROTECTED]:5060;tag=1 To: sut sip:[EMAIL PROTECTED]:5060;tag=as3e7533a6 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ' . 2005-02-08 00:23:36: Unexpected message for Call-ID '[EMAIL PROTECTED]': while expecting '100' response, received 'SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060 From: sipp sip:[EMAIL PROTECTED]:5060;tag=2 To: sut sip:[EMAIL PROTECTED]:5060;tag=as43cce205 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ 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 signature.asc Description: OpenPGP digital signature ___ 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] jitterbuffers - suggested settings
I recommend to deactivate the current jitter buffer and wait till a new one is ready. Joachim. Stuart Elvish wrote: Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know if anybody could suggest the number to put in jitterbuffers= and whether or not they have found this to affect the echo. Any suggestions will be greatly appreciated. Kind Regards Stuart ___ 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 signature.asc Description: OpenPGP digital signature ___ 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] SIP port blocked in Dubai ?
Hello, We had same problem in other african country. We could resolve it through using IAX Bridge in Asterisk since it only uses one port of yoru choice. For your solution, you need: 1-) Scan outgoing / incoming open ports by your ISP; 2-) If there remains many open ports, you may still run SIP by changing ports; 3-) Alternately, you need to get IAX bridge working if need be; It is my understanding the Government considers VoIP illegal since it would compete with the Government run service in Dubai. This is based on a conversation with a coworker who was raised there and just returned from a vacation home. You should check this out to make sure my facts are correct. Manjit Riat wrote: Does any know if SIP ports are blocked in dubai (UAE)? Anyone in UAE using FWD or similar services and connecting to SIP proxies in US? Thanks. This message was sent using IMP, the Internet Messaging Program. ___ 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] Autodetecting faxes
Howard Lowndes wrote: Apologies. I meant You do have a context called Fax...? I don't. And it's working absolutely fine. I have SpanDSP installed, faxdetect=both and the FaxReceive macro is shamelessly lifted from http://www.voip-info.org/wiki-Asterisk+fax -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ 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] Autodetecting faxes
Michael Welter wrote: Changing the order of things in extensions.conf around a smidge got it all working nicely :- [inbound-from-pstn] include = default exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,Playback(thank-you-for-calling-please-wait-a-moment) exten = fax,1,Macro(faxreceive) exten = s,4,Do the normal phone call gubbins Is the position of the fax extension, between priorities 3 and 4, significant? What does 'show dialplan' display for the fax extension? It's there as much for flow readability as anything... The change of order was as much referring to moving the Playback forward from the voice handling macro, to give * time to hear the fax beep. Show Dialplan gives :- In each of my inbound call contexts 'fax' = 1. Macro(faxreceive) [pbx_config] No other mention at all. -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ 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] Asterisk and Sipgate problem...
Hello all. I'm having an odd problem getting * and sipgate to work
together. From Sipgate support I have gotten this repsonse to my query:
=
Your Asterisk is registering incorrectly with our servers. It registers
like this: sip:[EMAIL PROTECTED]:5076
The s should be your SIP ID. Anything else is rejected. I don't know
where you can find this setting, but from our perspective that is where
the problem is. If you find it, please let me know.
=
If there is anyone that can shed some light on this odd problem it would
be greatly appreciated. If more info is needed please ask.
My configs look like this:
SIP.CONF:
register = XXX:[EMAIL PROTECTED]
[sipgate]
type=peer
context=in_sipgate
username=XXX
secret=
auth=md5
host=sipgate.co.uk
disallow=all
allow=ulaw
insecure=very
context=in_sipgate
canreinvite=no
fromuse=XXX
fromdomain=sipgate.co.uk
EXTENSIONS.CONF:
[in_sipgate]
exten = s,1,Answer
exten = s,2,Dial(SIP/44|30|t)
exten = s,3,Voicemail(u44)
[out_sipgate]
exten = _8.,1,SetCallerID(02070zzz)
exten = _8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]|90|r)
exten = _8.,3,Busy
exten = _8.,4,Hangup
--
Robert P. McKenzie | GammaRay Technical Services Ltd
[EMAIL PROTECTED] | [EMAIL PROTECTED]
http://www.uk-experience.com | http://www.gammaray-tech.com
Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014
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[Asterisk-Users] MD5 in SIP's register = ...
Hello Everyone! I just want to make sure if such a mess could work for sip channel: In sip.conf: ; register = some_md5_checksum@host ; ; [host] hostname=some_address auth=md5 Greets Tomek ___ 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] CVS or release?
hi is the v1-0 CVS branch supposed to be stable as in STABLE, or should one use releases? roy ___ 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] Question about TDM11B Configuration
Hello all, iwould liketo configure TDM11B with Asterisk, if any one have the configuration steps please provide me it. 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] snom soft phone
No, is default for snom phones. Sven On Tuesday 08 February 2005 08:37, Altus Snyman wrote: Did you try 00 That is what it is on the 220 On Tue, 2005-02-08 at 09:36, Paradise Dove wrote: what is the password for Administrator in the softphone? On Tue, 8 Feb 2005 08:01:07 +0100, Christian Stredicke [EMAIL PROTECTED] wrote: Go to the web page, in Preferences there are two pull down menus for Audio Input and Autio Output. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P. Sent: Tuesday, February 08, 2005 2:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] snom soft phone Hi, How do I change the default audio device ? I have one of those USB headset (which actually is another soundcard) but the simulation insist in using my Soundblaster Live card :( -- Juanjo sin .sig :( ___ 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 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 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 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 -- --- See our FAQs at: http://www.snom.com/faq0.html?L=1 Whitepapers at: http://www.snom.com/white_papers.html --- snom technology AG Pascalstraße 10b D-10587 Berlin Sven Fischer fax +49 30 39833111 mailto:[EMAIL PROTECTED] http://www.snom.comsip:[EMAIL PROTECTED] --- ___ 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] Zaptel down after upgrade.
Ok, When I tried newer version of zaptel, libpri, asterisk, it didn't work. My spans get down and RED. Then, I tried to go back to previous version. No way. Now, my E1 are up again, thanks for all your advice. I don't know which one solve my problem, so, here's what I've done. To get back to my previous version, in order : - I reboot to clean up zaptel device loaded in memory - I delete /usr/lib/asterisk/modules/* to remove asterisk modules that have been installed with the newer version and which is not supported by the older. - I reinstall all old library in this order : zaptel, libpri, and then asterisk I reboot again, but my 4 E1 were still RED. Then, I called my E1 supplier and ask them what they seen. My E1 was locked in their side. Too much alarm They unlock and reinitialise them. Now, it's ok again. Thanks again, I learned lot of things during this problem. Regis -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de tim panton Envoyé : lundi 7 février 2005 21:29 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Zaptel down after upgrade. On 7 Feb 2005, at 20:17, Régis MARTIN wrote: I tried to reinstall all previous version (zapata, zaptel, libpri and asterisk) I reboot. And then... same thing :( Ring your T1 supplier, and ask them what they see. They may well have marked it as out-of-service, in which case it won't come back 'till they re-enable it. Tim. http://www.westhawk.co.uk/ ___ 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 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] Help on Load Testing
Can anybody help me in sipp for load testing on asterisk? How to use sipp with asterisk?? Thanks RegardsRitesh Jalan ___ 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] Autodetecting faxes
That's all very well, but what do you do if you only have SIP extensions and IAX trunk - no Zaptel card. Will Fax detection still work at all? Thanks Mike - Original Message - From: Adrian Chapman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 08, 2005 8:24 PM Subject: Re: [Asterisk-Users] Autodetecting faxes Michael Welter wrote: Changing the order of things in extensions.conf around a smidge got it all working nicely :- [inbound-from-pstn] include = default exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,Playback(thank-you-for-calling-please-wait-a-moment) exten = fax,1,Macro(faxreceive) exten = s,4,Do the normal phone call gubbins Is the position of the fax extension, between priorities 3 and 4, significant? What does 'show dialplan' display for the fax extension? It's there as much for flow readability as anything... The change of order was as much referring to moving the Playback forward from the voice handling macro, to give * time to hear the fax beep. Show Dialplan gives :- In each of my inbound call contexts 'fax' = 1. Macro(faxreceive) [pbx_config] No other mention at all. -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.6 - Release Date: 7/02/2005 ___ 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] Registering Microsoft RTC Client API SDK with Asterisk
Last week I asked how to register the Microsoft Real_time Communications
Client with Asterisk. No replys came, however I managed to figure it out
myself. I thought I'd just post the solution for anyone else in the
future wanting to do the same.
Regards
Jerry
CString gXMLProfile = \
provision key=\5B29C449-29EE-4fd8-9E3F-04AED077690E\\n\
name=\Asterisk\\n\
\n\
user account=\username\\n\
uri=\username\\
/\n\
\n\
sipsrv addr=\Asterisk_Address\\n\
protocol=\udp\\n\
auth=\digest\\n\
role=\registrar\\n\
session party=\first\ type=\pc2ph\ /\n\
/sipsrv\n\
\n\
/provision\n;
IRTCClient2 *g_pClient = NULL;
BSTR bstrXMLProfile;
HRESULT hr = E_FAIL;
// initialize COM.
hr = CoInitialize(NULL);
if (FAILED(hr))
{
TRACE(CoInitialize failed:hr=%x\n, hr);
return;
}
// Create the RTC client
hr = CoCreateInstance(
__uuidof(RTCClient),
NULL,
CLSCTX_INPROC_SERVER,
__uuidof(IRTCClient2),
(LPVOID *)g_pClient
);
if (FAILED(hr))
{
TRACE(CoCreateInstance failed: hr=%x\n, hr);
return;
}
hr = g_pClient-Initialize();
if (hr != S_OK)
TRACE(Unable to initialise\n);
else
{
TRACE(Hello, RTC!\n);
TRACE(\nXML Schema = \n%s\n, gXMLProfile);
IRTCClientProvisioning *pIRTCClientProvisioning = NULL;
IRTCProfile*pIRTCProfile= NULL;
bstrXMLProfile= gXMLProfile.AllocSysString();
// Perform QI for the Provisioning interface.
hr = g_pClient-QueryInterface(IID_IRTCClientProvisioning,
reinterpret_castvoid **(pIRTCClientProvisioning));
// If (hr != S_OK), process the error here.
if (hr != S_OK)
TRACE(QueryInterface failed\n);
else
{
// Create the Profile object.
hr = pIRTCClientProvisioning-CreateProfile(bstrXMLProfile,
pIRTCProfile);
// If (hr != S_OK), process the error here.
if (hr != S_OK)
TRACE(CreateProfile failed 0x%X\n, hr);
else
{
// Enable the Profile and Register.
hr =
pIRTCClientProvisioning-EnableProfile(pIRTCProfile, RTCRF_REGISTER_ALL);
// If (hr != S_OK), process the error here.
if (hr != S_OK)
{
TRACE(Enable Profile failed\n);
}
else
{
TRACE(ALL OK\n);
}
}
}
}
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Re: [Asterisk-Users] Best OS for Asterisk--newbie!!!
Thankyou so much Chris and Roger, I really appreciate your response and suggestions good luck :)) kind regards Siju ___ 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] DTMF CLIP in Sweden and others
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Are this currently working with CVS-HEAD? I've got an X100P-clone, and I've patched the zaptel drivers. But the Asterisk patches seems to be there. But I can't make it receive Caller-ID! Btw, by doing a cvs checkout asterisk, the HEAD-version will be downloaded right? Thanks! -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCCJrL/4dZjWjLCy0RAglwAJ0Tu9dQRDy9XCanXCeTiJJd4zS4NACcDBYq 6JVkIIj+wDu+7drgxjQ73U0= =LiqB -END PGP SIGNATURE- ___ 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] Asterisk and Sipgate problem...
I saw at least three details in your config, which could result in problems.
Since I'm relative new to asterisk, take my tips with care.
register = XXX:[EMAIL PROTECTED]
should be register = XXX:[EMAIL PROTECTED]/XXX
fromuse=XXX
should be fromuser=XXX
auth=md5 ; I'm not shure if this works..., perhaps disable it in the first
step
Here a small example that works for us:
sip.conf:
[general]
realm = hallinux2.gwsnettech.local
port = 5060
bindaddr = 0.0.0.0
context = default
disallow=all
allow=alaw
allow=ulaw
allow=gsm
register = 081503:[EMAIL PROTECTED]/081503
language=de
tos=0x04
[sipgate]
type=friend
username=081503
secret=xx
host=sipgate.de
fromuser=081503
fromdomain=sipgate.de
nat=yes
context=incomingsipgate
context=default
canreinvite=yes
insecure=very
extensions.conf:
[sipgate]
include = default
exten = _9*.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _9*.,2,Congestion
exten = _9*.,102,Busy
[incomingsipgate]
exten = 5339504,1,Dial(SIP/6301,10,tr)
exten = 5339504,2,Dial(SIP/6301,10,tr)
exten = 5339504,3,SetLanguage(de)
exten = 5339504,4,Voicemail2(6301)
exten = 5339504,5,Hangup
Hope this helps a little bit...
Guido Hecken
Hello all. I'm having an odd problem getting * and sipgate to work
together. From Sipgate support I have gotten this repsonse to my query:
=
Your Asterisk is registering incorrectly with our servers. It registers
like this: sip:[EMAIL PROTECTED]:5076
The s should be your SIP ID. Anything else is rejected. I don't know
where you can find this setting, but from our perspective that is where
the problem is. If you find it, please let me know.
=
If there is anyone that can shed some light on this odd problem it would
be greatly appreciated. If more info is needed please ask.
My configs look like this:
SIP.CONF:
register = XXX:[EMAIL PROTECTED]
[sipgate]
type=peer
context=in_sipgate
username=XXX
secret=
auth=md5
host=sipgate.co.uk
disallow=all
allow=ulaw
insecure=very
context=in_sipgate
canreinvite=no
fromuse=XXX
fromdomain=sipgate.co.uk
EXTENSIONS.CONF:
[in_sipgate]
exten = s,1,Answer
exten = s,2,Dial(SIP/44|30|t)
exten = s,3,Voicemail(u44)
[out_sipgate]
exten = _8.,1,SetCallerID(02070zzz)
exten = _8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]|90|r)
exten = _8.,3,Busy
exten = _8.,4,Hangup
--
Robert P. McKenzie | GammaRay Technical Services Ltd
[EMAIL PROTECTED] | [EMAIL PROTECTED]
http://www.uk-experience.com | http://www.gammaray-tech.com
Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014
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[Asterisk-Users] SIP jitter?
hi how can I tune SIP jitter? is it possible today in asterisk? ryo ___ 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] Asterisk and Sipgate problem...
Thanks for your reply. I had tried the register line as you have it
before and it fails. When I call my UK number from sipgate it's just a
busy signal. If I remove my SIP-ID as I have in my current configs the
calls go through fine. Outgoing calls another matter. It seems to lock
up my router and I can't make calls.
A question about your config.. in your extensions.conf you have in your
incoming section:
exten = 5339504,1,Dial(SIP/6301,10,tr)
Where does the number 5339504 come from? Sipgate has told me that I
should be putting my SIP-ID there.. but it also fails.
So far Sipgate has proved to be the most problematic provider I've tried
using.. and now they have my money with no refunds and I can't use them.
Sigh.
Hecken, Guido wrote:
I saw at least three details in your config, which could result in problems.
Since I'm relative new to asterisk, take my tips with care.
register = XXX:[EMAIL PROTECTED]
should be register = XXX:[EMAIL PROTECTED]/XXX
fromuse=XXX
should be fromuser=XXX
auth=md5 ; I'm not shure if this works..., perhaps disable it in the first
step
Here a small example that works for us:
sip.conf:
[general]
realm = hallinux2.gwsnettech.local
port = 5060
bindaddr = 0.0.0.0
context = default
disallow=all
allow=alaw
allow=ulaw
allow=gsm
register = 081503:[EMAIL PROTECTED]/081503
language=de
tos=0x04
[sipgate]
type=friend
username=081503
secret=xx
host=sipgate.de
fromuser=081503
fromdomain=sipgate.de
nat=yes
context=incomingsipgate
context=default
canreinvite=yes
insecure=very
extensions.conf:
[sipgate]
include = default
exten = _9*.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _9*.,2,Congestion
exten = _9*.,102,Busy
[incomingsipgate]
exten = 5339504,1,Dial(SIP/6301,10,tr)
exten = 5339504,2,Dial(SIP/6301,10,tr)
exten = 5339504,3,SetLanguage(de)
exten = 5339504,4,Voicemail2(6301)
exten = 5339504,5,Hangup
Hope this helps a little bit...
Guido Hecken
Hello all. I'm having an odd problem getting * and sipgate to work
together. From Sipgate support I have gotten this repsonse to my query:
=
Your Asterisk is registering incorrectly with our servers. It registers
like this: sip:[EMAIL PROTECTED]:5076
The s should be your SIP ID. Anything else is rejected. I don't know
where you can find this setting, but from our perspective that is where
the problem is. If you find it, please let me know.
=
If there is anyone that can shed some light on this odd problem it would
be greatly appreciated. If more info is needed please ask.
My configs look like this:
SIP.CONF:
register = XXX:[EMAIL PROTECTED]
[sipgate]
type=peer
context=in_sipgate
username=XXX
secret=
auth=md5
host=sipgate.co.uk
disallow=all
allow=ulaw
insecure=very
context=in_sipgate
canreinvite=no
fromuse=XXX
fromdomain=sipgate.co.uk
EXTENSIONS.CONF:
[in_sipgate]
exten = s,1,Answer
exten = s,2,Dial(SIP/44|30|t)
exten = s,3,Voicemail(u44)
[out_sipgate]
exten = _8.,1,SetCallerID(02070zzz)
exten = _8.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]|90|r)
exten = _8.,3,Busy
exten = _8.,4,Hangup
--
Robert P. McKenzie | GammaRay Technical Services Ltd
[EMAIL PROTECTED] | [EMAIL PROTECTED]
http://www.uk-experience.com | http://www.gammaray-tech.com
Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014
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Robert P. McKenzie | GammaRay Technical Services Ltd
[EMAIL PROTECTED] | [EMAIL PROTECTED]
http://www.uk-experience.com | http://www.gammaray-tech.com
Ecademy Profile: http://www.ecademy.com/account.php?op=viewid=64014
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[Asterisk-Users] Voicemail not working properly
i am working on asterisk. i am using fedora core 2 on my asterisk mechine. when i was working on stable version my voicemailmenu was working well. i can lissten to menu and send dtmf to control menu now i have compiled CVS version of asterisk. now when i configure my voicemail for any extension suppose i declared a voicemail box for user 3000. when i dial to 3000 i cannot have any menu there is no voicemail working. i have 3001 for voicemailmenu as well it is not working. is there any problem in CVS version. i am working on real time mysql addon with asterisk. thanks __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.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] high-quality, high-bandwidth codecs?
hi are there any codecs around that allows high quality as in studio lite? it may consume high bandwidth, and hopefully allow some packet loss. roy ___ 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] jitterbuffers - suggested settings
Any idea when that is likely to be ready? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of joachim Sent: Tuesday, February 08, 2005 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] jitterbuffers - suggested settings I recommend to deactivate the current jitter buffer and wait till a new one is ready. Joachim. ___ 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] live monitoring (SIP only)
Hi, is it and how is it possible to live monitor (barge - in) a SIP to SIP call without any Zap Interface? I am using asterisk 1.0.5 with chan_capi from Junghanns and SIP clients. I was looking for chan_spy application but it seems to be no longer available. Bye, Sven___ 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] How to xfer calls or is my setup wrong?
I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss... The call comes in from an external number via IAX2 (0870xxx) which I can answer on any of the ringing extensions no problem. But when I need to xfer that call I am more or less stuck. I have read various posts and something about *8# ? seemed to partially work one on the grandstream but I haven't been able to reproduce that... The CLI doesn't show anything odd... Any ideas? ___ 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] live monitoring (SIP only)
Hello, is it and how is it possible to live monitor (barge - in) a SIP to SIP call without any Zap Interface? I am using asterisk 1.0.5 with chan_capi from Junghanns and SIP clients. I was looking for chan_spy application but it seems to be no longer available. You can do something like this with the Flash Operator Panel ( http://www.asternic.org ). chan_spy would be a better option because you can use it from the dialplan. As a workaraound, FOP lets you drag your phone to a bridged call and put the three in a meetme room, with the option to start the 3rd led muted so the other won't notice the interruption. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] DTMF CLIP in Sweden and others
On Tue, 8 Feb 2005, Daniel Nyström wrote: Are this currently working with CVS-HEAD? I've got an X100P-clone, and I've patched the zaptel drivers. But the Asterisk patches seems to be there. But I can't make it receive Caller-ID! The X100P is unsuited for use with the Swedish PSTN for several reasons - wrong line impedance, no polarity reversal sensing etc. Peter ___ 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] SIP jitter?
On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote: how can I tune SIP jitter? is it possible today in asterisk? I assume you are asking for how to alleviate the effects of jitter on the RTP audio streams initated by SIP? Asterisk currently only has a jitter buffer for IAX, not for RTP streams. There are pland for the next generation jitter buffer code to hook into RTP as well. There is an entry on the bug tracker that touches on this topic. Peter ___ 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] How to xfer calls or is my setup wrong?
What asterisk version I know we had a problem with one of the cvs We couldn't use the transfer buttons,but # worked What about the Dail(SIP/111,12,tT) in your extensions.conf On Tue, 2005-02-08 at 13:50, Mark Benson wrote: I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss... The call comes in from an external number via IAX2 (0870xxx) which I can answer on any of the ringing extensions no problem. But when I need to xfer that call I am more or less stuck. I have read various posts and something about *8# ? seemed to partially work one on the grandstream but I haven't been able to reproduce that... The CLI doesn't show anything odd... Any ideas? ___ 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 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
Antwort: Re: [Asterisk-Users] live monitoring (SIP only)
Yes, that would work - but I have no Zap and therefor no meetme - or is there a way to start meetme with SIP interfaces only ? [EMAIL PROTECTED] schrieb am 08.02.2005 08:53:06: Hello, is it and how is it possible to live monitor (barge - in) a SIP to SIP call without any Zap Interface? I am using asterisk 1.0.5 with chan_capi from Junghanns and SIP clients. I was looking for chan_spy application but it seems to be no longer available. You can do something like this with the Flash Operator Panel ( http://www.asternic.org ). chan_spy would be a better option because you can use it from the dialplan. As a workaraound, FOP lets you drag your phone to a bridged call and put the three in a meetme room, with the option to start the 3rd led muted so the other won't notice the interruption. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ 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 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] live monitoring (SIP only)
On 08/02/2005 19:23 [EMAIL PROTECTED] said the following: and SIP clients. I was looking for chan_spy application but it seems to be no longer available. oddly, ChanSpy seems to be removed from mantis. any idea why this was done ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ 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: Antwort: Re: [Asterisk-Users] live monitoring (SIP only)
On Tue, 8 Feb 2005, Sven Lohmann wrote: Yes, that would work - but I have no Zap and therefor no meetme - or is there a way to start meetme with SIP interfaces only ? Use ztdummy or zaprtc. All that is needed is the zaptel timing. Another option may be to use app_conference (use google). Peter ___ 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] live monitoring (SIP only)
I've asked the same question at least three times, but no-one has replied. Surely the person who removed the bug must know ;) Julian. Dinesh Nair wrote: On 08/02/2005 19:23 [EMAIL PROTECTED] said the following: and SIP clients. I was looking for chan_spy application but it seems to be no longer available. oddly, ChanSpy seems to be removed from mantis. any idea why this was done ? ___ 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
Antwort: Re: Antwort: Re: [Asterisk-Users] live monitoring (SIP only)
I am one of these unhappy people using the wrong USB chip and building my own kernel (RTC is activated) is no option due to company policies. [EMAIL PROTECTED] schrieb am 08.02.2005 13:11:44: On Tue, 8 Feb 2005, Sven Lohmann wrote: Yes, that would work - but I have no Zap and therefor no meetme - or is there a way to start meetme with SIP interfaces only ? Use ztdummy or zaprtc. All that is needed is the zaptel timing. Another option may be to use app_conference (use google). Peter ___ 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 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] spandsp
Good day all I have a asterisk installation,1.0.3, and spandsp. I got asterisk working,I edited the make file myself. Now when I receive a fax I only get half a page or nothing any Ideas why Please let me know Altus ___ 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] jitterbuffers - suggested settings
On February 8, 2005 03:31 am, joachim wrote: I recommend to deactivate the current jitter buffer and wait till a new one is ready. Any particular reason why? I am using the following jitter buffer settings with Jan 2005 CVS HEAD without any issues, and it seems to work reasonably well: jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=100 minexcessbuffer=50 jittershrinkrate=1 -A. ___ 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] Snom programmable leds / keys usage for pickup groups?
Would it be possible to use the programmable led+keys on the Snom phones to signal that there is an incoming call that is ringing a call group or pickup group? We use this on our existing PBX if for example the accounting dept. is out for lunch but nobody can hear their phones. This way you can see an incoming call (and we hate voicemail) :) Thanks! ___ 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] Snom programmable leds / keys usage for pickup groups?
On Tue, Feb 08, 2005 at 01:47:57PM +0100, Remco Barende wrote: Would it be possible to use the programmable led+keys on the Snom phones to signal that there is an incoming call that is ringing a call group or pickup group? We use this on our existing PBX if for example the accounting dept. is out for lunch but nobody can hear their phones. This way you can see an incoming call (and we hate voicemail) :) I'm also curious about how configurable the Snom's buttons are. Can they be assigned, say SIP/1, SIP/2, SIP/3, etc and light up when that channel is in use? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Re: Asterisk and Sipgate problem...
Robert P. McKenzie schrieb:
So far Sipgate has proved to be the most problematic provider I've tried
using.. and now they have my money with no refunds and I can't use them.
Sigh.
I never had any problems with (the german) sipgate so far.
My sip.conf (only the sipgate-parts):
register = 0815:[EMAIL PROTECTED]/0815
context = incoming
[sipgate]
type=friend
username=0815
host=sipgate.de
fromuser=0815
fromdomain=sipgate.de
nat=no
canreinvite=no
-
extensions.conf:
[incoming]
exten = _X.,1,Dial(Phone/phone0,90)
exten = _X.,2,Hangup
-
callout.php: (I'm using an AGI-script for dialing)
function DialSipgate($agi, $number)
{
$agi-conlog(Dial Sipgate);
$agi-set_callerid(Michael Vogel 0815);
$agi-agi_exec(EXEC DIAL SIP/.$number.@sipgate);
}
-
Bye!
Michael
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RE: [Asterisk-Users] jitterbuffers - suggested settings
? What's wrong with the current jitterbuffer.. -Original Message- From: joachim [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 08, 2005 2:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] jitterbuffers - suggested settings I recommend to deactivate the current jitter buffer and wait till a new one is ready. Joachim. Stuart Elvish wrote: Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know if anybody could suggest the number to put in jitterbuffers= and whether or not they have found this to affect the echo. Any suggestions will be greatly appreciated. Kind Regards Stuart ___ 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 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 timeouts after 30sec's everytime.
On Mon, Feb 07, 2005 at 11:08:51PM -0500, Jon Radon wrote: Instead of hijacking the thread you could just look it up. (HINT: it's a feature in cvs) I'm using stable rather than CVS. I did look on voip-info and I searched the mailing list archives. If there's another place I could've looked before asking, I'd love to know it to save redundant questions here in the future. Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] high-quality, high-bandwidth codecs?
On Tue, Feb 08, 2005 at 12:26:39PM +0100, Roy Sigurd Karlsbakk wrote: hi are there any codecs around that allows high quality as in studio lite? it may consume high bandwidth, and hopefully allow some packet loss. I'm not sure what studio lite means to you. Maybe hard figures would be more precise. G.722 might be interesting : 64 kbps, 7 kHz. It's not free. Otherwise, MP3 or OGG might be ok ? -- Nicolas Bougues ___ 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] Q: ISDN / E1-PRI - fax problems - Receiving and setting of Service Indicator (SIN) / Bearer Capability (BC) / High Level Compatibility (HLC) / Low Level Compatibility (LLC)
hi, i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. The problem is, that the Bearer Capability (BC) together with the High Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the Service Indicator (SIN). The SIN is used to determine if the call is voice, fax or data. It's essential to set the SIN so the called party is able which device has to answer a call (e.g. telephone or fax) as far as i dug into the source neither the BC nor the HLC or LLC data is forwarded to a dialplan variable and only the BC is decoded in libpri. has anyone a solution for this? is there any usable documentation on the HLC or LLC octets (bytes)? i searched etsi and was overwhelmed with the searchresults (1531). what i need to modify libpri would be a table of possible values and where to find the HLC and LLC fields in the D-Channel. regards frank ___ 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] Autodetecting faxes
Mike Sander wrote: That's all very well, but what do you do if you only have SIP extensions and IAX trunk - no Zaptel card. Will Fax detection still work at all? Absolutely no idea. We're working with X100s, and haven't started looking at Fax sending yet. -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ 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] jitterbuffers - suggested settings
Apparantly the new one will do things like interpolation so that if packets are lost it will generate new ones to fill the gap. The current jitterbuffer doesn't do that so you get silence on packet loss. There are a bunch of other features too that I don't remember, but that was the most interesting to me last time I looked. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Schulte Sent: 08 February 2005 13:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] jitterbuffers - suggested settings ? What's wrong with the current jitterbuffer.. -Original Message- From: joachim [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 08, 2005 2:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] jitterbuffers - suggested settings I recommend to deactivate the current jitter buffer and wait till a new one is ready. Joachim. Stuart Elvish wrote: Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know if anybody could suggest the number to put in jitterbuffers= and whether or not they have found this to affect the echo. Any suggestions will be greatly appreciated. Kind Regards Stuart ___ 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 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 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] high-quality, high-bandwidth codecs?
Nicolas Bougues wrote: On Tue, Feb 08, 2005 at 12:26:39PM +0100, Roy Sigurd Karlsbakk wrote: hi are there any codecs around that allows high quality as in studio lite? it may consume high bandwidth, and hopefully allow some packet loss. I'm not sure what studio lite means to you. Maybe hard figures would be more precise. G.722 might be interesting : 64 kbps, 7 kHz. It's not free. Otherwise, MP3 or OGG might be ok ? G.722 is free, as any patents have expired. G.722.1 and G.722.2 are not free. They are *much* better, though. G.722 gives quite poor performance. I'm not sure if that is because of poor design, and a desire to keep its compute needs in check. Whatever the history, looked at now it sucks. Vorbis or speex would be much better. Regards, Steve ___ 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] Re: Voicemail not working properly
can anyone tell me how to add extension to extension table i think this is the main prblem. any one to guide me. ++-+-+--+--+-+ | id | context | exten | priority | app | appdata | ++-+-+--+--+-+ | 1 | default | 3000|1 | Dial(SIP/3000,20,tr) | | | 2 | default | 3000|2 | VoiceMail | u | | 3 | default | 3000| 102 | VoiceMail | b | | 4 | default | 3001|1 | Ringing | | | 5 | default | 3001|2 | Wait(2) | | | 6 | default | 3001|3 | VoicemailMain| | | 7 | default | _574555 |1 | Wait | 2 | | 8 | default | _574555 |2 | SayNumber | 102 | ++-+-+--+--+-+ __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ 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] how to pop up called number details using php scripts in agi scripts
Hi to all, I and using asterisk with following setup. 1. TDM400p card with four FXS modules, so there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 I want your guidance for the following issue. with help of agi scripts i am able to insert caller id number in database of mysql now i want to pop it up via html or php page but can any one of you let me know how can i use php scripts in agi scipts so that i can pick called id number from database and retreive the caller record from database using this called id number cheers mazhar ___ 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] VoIP extn number planning
Looking for some advanced thoughts relative to exten number assignments. We're in the planning stage for rolling out asterisk at multiple small US telco/isp operations. Their typical voip customer has had their pstn line for a long time and wants to keep the pstn line and number, but add voip to their existing home/soho arrangement. One approach (from a planning perspective) is to deploy spa3k's at the customer's location and configure it for pstn ring thru to line1, implementing something like dial-8 for outbound voip calls, and defaulting all other outbound calls to the pstn line. The arrangement addresses 911 calls, etc, nicely. However for inbound voip calls, there does not seem to be any industry planning for addressing number assignments associated with voip facilities. (With 20+ years of telephony engineering experience, I do understand the value of ss7, number portability, dundi, etc.) On the surface there seems to be two basic approaches for these customers: 1. If the customer's current pstn number is 402-234-5678, assign the same number to their voip facility, or, 2. For each Central Office (in this case), assign 100 or 1000 local exchange number (from their existing pstn numbering plan) to voip customers, or, 3. Assign some random number (something like FWD's approach) and translate that number within the voip switching facility (eg, asterisk in this case). The voip number that is assigned doesn't make a lot of difference today as the current spotty/independent voip-system implementations really don't address voip call completion on a larger/nationwide voip scale (other than through existing pstn facilties). The approach in #1 would suggest that each remote voip system must do some sort of lookup on a per-number basis for call completion routing. Keep in mind the exact same number is accessible via the existing pstn network or via a voip network. The approach in #2 would suggest remote translations might be easier but probably does not support number portability in the long run. The approach in #3 is certainly workable, but probably involves a number change in the long term, disruption to existing customers, and in some cases a truck-roll to facilitate the change. Not all that cool and probably the highest cost in the long run. Has anyone on this list given any serious thought on how to approach this topic for the longer run? Rich ___ 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] Bristuff - analogue communication over ISDN
Hello, I have * config with one quadBRI card for PSTN ISDN lines and one TDM400P for analogue faxes and modems. I could never get a fax or modem to work reliably over ISDN. With bristuff 0.2.0 RC3a (* 1.0.3) modem connection would drop after a few secs, and fax would never get through if it had more than two pages. Then I upgraded to bristuff 0.2.0 RC6 (* 1.0.5) and things got better, but hardly satisfactory. Modem connection holds up to 15 min, and faxes go through although output is sometimes garbled. I would appreciate any pointer on how to begin resolving this issue. I realize that the best advice is to get rid of analogue technology alltogether, but unfortunately that is currently not an option. My zaptel.conf and zapata.conf are attached. Niksa [trunkgroups] [channels] language=en callwaiting=yes callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes hidecallerid=no useincomingcalleridonzaptransfer=yes relaxdtmf=no echotraining=no echocancel=no echocancelwhenbridged=no rxgain=0.0 txgain=0.0 callgroup=0 pickupgroup=0 context=from-tcom switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes immediate=no signalling=bri_cpe group=1 channel = 1,2,4,5 context=from-mobile group=2 channel = 7,8 context=international signalling=bri_net_ptmp callreturn=yes callerid=Servis ISDN 36 group=3 channel = 10,11 signalling=fxo_ks busydetect=yes busycount=4 callerid=Fax modem 22 group=4 channel = 13 callerid=Servis analog 24 group=5 channel = 14 callerid=Dragan modem 26 group=6 channel = 15 callerid=Fax 27 group=7 channel = 16 span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1-2 dchan=3 bchan=4-5 dchan=6 bchan=7-8 dchan=9 bchan=10-11 dchan=12 fxoks=13 fxoks=14 fxoks=15 fxoks=16 loadzone=nl defaultzone=nl ___ 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
Fw: [Asterisk-Users] Help on Load Testing
Thanks RegardsRitesh JalanSenior Engineer - Test AuditNet4india Ltd.703, Bhikaji Cama Bhawan11, Bhikaji Cama PlaceNew Delhi 110066Tel: 91 (011) (26160129 - 131) (Extn 131)URL: http://www.net4india.com - Original Message - From: Ritesh Jalan To: asterisk-users@lists.digium.com Sent: Tuesday, February 08, 2005 3:47 PM Subject: [Asterisk-Users] Help on Load Testing Can anybody help me in sipp for load testing on asterisk? How to use sipp with asterisk?? Thanks RegardsRitesh Jalan ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] CVS or release?
On Tue, 8 Feb 2005 10:47:40 +0100, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: is the v1-0 CVS branch supposed to be stable as in STABLE, or should one use releases? v1-0 is the tag used for the latest changes to the stable branch. Releases are still your best bet, but if you are monitoring the CVS mailing list for commits to v1-0 stable, then you may see a patch go in that fixes some bug you've been having. The rule generally is don't run CVS unless you're monitoring the CVS mailing list. -rv1-0-5 will get you version 1.0.5 of Asterisk, which is the latest released version. Thanks, Leif Madsen. http://www.leifmadsen.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] SIPP load testing - unexpected message - anyone using sipp sucessfully ?
On Tue, 08 Feb 2005 10:16:46 +0200, joachim [EMAIL PROTECTED] wrote: SIPP works for asterisk testing too, but you need the correct commandline. What did you use ? Perhaps you can just give us the _correct_ command line for those of us who are unknowing? :) Thanks, Leif Madsen. http://www.leifmadsen.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] Q: ISDN / E1-PRI - fax problems - Receiving and setting of Service Indicator (SIN) / Bearer Capability (BC) / High Level Compatibility (HLC) / Low Level Compatibility (LLC)
Yeah, I also noticed that * lacks the ability to forward calls based on the type of call, but I have no idea whether this issue has any priority with the development team. It is probably better to ask this question on Asterisk-Dev mailing list. Frank Sautter wrote: hi, i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. The problem is, that the Bearer Capability (BC) together with the High Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the Service Indicator (SIN). The SIN is used to determine if the call is voice, fax or data. It's essential to set the SIN so the called party is able which device has to answer a call (e.g. telephone or fax) as far as i dug into the source neither the BC nor the HLC or LLC data is forwarded to a dialplan variable and only the BC is decoded in libpri. has anyone a solution for this? is there any usable documentation on the HLC or LLC octets (bytes)? i searched etsi and was overwhelmed with the searchresults (1531). what i need to modify libpri would be a table of possible values and where to find the HLC and LLC fields in the D-Channel. regards frank ___ 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 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
-Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the phone at the other end being hung up, followed immediately by a * touchtone. Then there's silence until I hang up. ___ 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] high-quality, high-bandwidth codecs?
are there any codecs around that allows high quality as in studio lite? it may consume high bandwidth, and hopefully allow some packet loss. I'm not sure what studio lite means to you. Maybe hard figures would be more precise. G.722 might be interesting : 64 kbps, 7 kHz. It's not free. Otherwise, MP3 or OGG might be ok ? Would it be hard to do a codec_ogg? roy ___ 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] Hangup detection with TDM400 in UK
Hi Try going into vi /etc/profile insert the lines in brackets. USER=`id -un` LOGNAME=$USER MAIL=/var/spool/mail/$USER MONITOR_EXEC=/usr/bin/soxmix VPB_TONE=BUSY,P,400,100,500(insert the following line) HOSTNAME=`/bin/hostname` HISTSIZE=1000 if [ -z $INPUTRC -a ! -f $HOME/.inputrc ]; then INPUTRC=/etc/inputrc fi export PATH USER LOGNAME MAIL HOSTNAME HISTSIZE INPUTRC MONITOR_EXEC VPB_TONE (and insert here VPB_TONE) == The 400 100 and 500 are related to your country use indications file for info on what those values should be. Regards Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Patrick Lidstone (Personal E-mail) Sent: Wednesday, February 02, 2005 2:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hangup detection with TDM400 in UK When a caller hangs up (e.g. after leaving a voicemail), my British Telecom exchange sends a continuous tone for about 15s and then silence. I can't get asterisk to recognise this tone as a hangup indication. I have tried indications.conf with both country=uk and country=us. My zapata.conf has busydetect=yes, callprogress=yes and I've tried setting busycount from 1 through 7 I am using kewlstart signalling on the FXO module. Any suggestions gratefully received - I really don't want to resort to using an absolute timeout. Thanks Patrick ___ 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 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 hardphone
Try ACT P104 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Monday, February 07, 2005 1:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] iax hardphone Hi! Is there such a beast yet available? - Digium IAXy - PA168 chipset: http://www.voip-info.org/tiki-index.php?page=PA168 - farfon (only test devices yet) - several products: http://www.iaxtalk.com/ Cheers, Philipp ___ 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 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] How to xfer calls or is my setup wrong?
I have this in my extensions.conf exten = 08700xx,1,Dial(SIP/test1SIP/test2SIP/test3,30,t) To ring a group of internal extensions for any call coming in on that number And exten = 100,1,Dial(SIP/test1,20,Trt) exten = 100,2,Voicemail(u100) exten = 100,3,Hangup() exten = 100,102,Voicemail(b100) exten = 100,103,Hangup() For each extension... Altus Snyman wrote: What asterisk version I know we had a problem with one of the cvs We couldn't use the transfer buttons,but # worked What about the Dail(SIP/111,12,tT) in your extensions.conf On Tue, 2005-02-08 at 13:50, Mark Benson wrote: I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss... The call comes in from an external number via IAX2 (0870xxx) which I can answer on any of the ringing extensions no problem. But when I need to xfer that call I am more or less stuck. I have read various posts and something about *8# ? seemed to partially work one on the grandstream but I haven't been able to reproduce that... The CLI doesn't show anything odd... Any ideas? ___ 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 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 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] Q: ISDN / E1-PRI - fax problems - Receiving and setting of Service Indicator (SIN) / Bearer Capability (BC) / High Level Compatibility (HLC) / Low Level Compatibility (LLC)
On Tue, 8 Feb 2005, Frank Sautter wrote: is there any usable documentation on the HLC or LLC octets (bytes)? i searched etsi and was overwhelmed with the searchresults (1531). what i need to modify libpri would be a table of possible values and where to find the HLC and LLC fields in the D-Channel. ITU q.931 clause 4.5.17 for HLC and 4.5.19 for LLC. I'd expect ETSI ETS 300 403 01 to include q.931 by reference, possibly noting some differences. Peter ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
On February 8, 2005 08:44 am, David Brodbeck wrote: The sequence I hear on the extension, when I plug in an analog phone, is the click of the phone at the other end being hung up, followed immediately by a * touchtone. Then there's silence until I hang up. Hmm... I bet it has everything to do with not having 't' or 'T' in the dialplan -- asterisk is ignoring the tones because it's potentially a security problem. At least that's my current working theory -- I am not sure if t/T listen ofr all DTMF or just #, and I also don't know the direction of your call (PBX - * or * - PBX). -A. ___ 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] how to pop up called number details using php scripts in agi scripts
On February 8, 2005 08:14 am, Mazhar Hussain wrote: I want your guidance for the following issue. with help of agi scripts i am able to insert caller id number in database of mysql now i want to pop it up via html or php page but can any one of you let me know how can i use php scripts in agi scipts so that i can pick called id number from database and retreive the caller record from database using this called id number I use Jabber and Perl for this: http://www.mixdown.ca/~andrew/astbot/ -A. ___ 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] Using a Dual WAN Load Balancing Device
We have a client that wants to bond 2 DSL circuits instead of getting a T-1 (or similar) at their office to run their VoIP traffic on. We came across this Multihomed Gateway (MH200): http://www.cyberpathinc.com/mh200/details.htm Does anybody think this would work if installed at the client location handling NAT for 10 Cisco 7960's and connecting to our public asterisk server? My concern (as is others on this list in regards to load balancing) is what would happen if a call had to be directed out the other WAN port of the MH200 or if a call were to come in on 1 circuit and it runs out of bandwidth - how would the call be delivered to the second circuit. Or even if during a call, the inbound audio is fine (since DSL usually has more bandwidth on the download), but the outbound audio stream had to be pushed out the other WAN port. Hope that all makes sense (I almost confused myself! LOL) I am not holding my breath that this is a viable solution, but was just wondering your thoughts. Thanks! Pedro ___ 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] How to xfer calls or is my setup wrong?
I put dtmfmode=rfc2388 into the sip.conf definitions for each sip client and now asterisk is recognising the # key press - guess it wasn't hearing the dtmf tones... Now blind xfer works - how do I do attended xfer? I have read posts about it being in the CVS version - I am running the 1.0.3 release... Altus Snyman wrote: What asterisk version I know we had a problem with one of the cvs We couldn't use the transfer buttons,but # worked What about the Dail(SIP/111,12,tT) in your extensions.conf On Tue, 2005-02-08 at 13:50, Mark Benson wrote: I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss... The call comes in from an external number via IAX2 (0870xxx) which I can answer on any of the ringing extensions no problem. But when I need to xfer that call I am more or less stuck. I have read various posts and something about *8# ? seemed to partially work one on the grandstream but I haven't been able to reproduce that... The CLI doesn't show anything odd... Any ideas? ___ 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 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 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
Fwd: [Asterisk-Users] Record() cut off after 40 sec
-- Forwarded message -- From: Carlos Gabriel Drach [EMAIL PROTECTED] Date: Tue, 8 Feb 2005 11:20:01 -0300 Subject: Re: [Asterisk-Users] Record() cut off after 40 sec To: Steven Critchfield [EMAIL PROTECTED] On Mon, 07 Feb 2005 15:35:46 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2005-02-07 at 18:25 -0300, Carlos Gabriel Drach wrote: On Mon, 07 Feb 2005 14:54:46 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: Looks like you are missing some information here. Specifically you don't have the portion that created the local channel. I am starting to wonder if your dial that creates the local channel isn't set for 40 seconds timeout. It would explain the hangup listed in the messages above. -- Steven Critchfield [EMAIL PROTECTED] Hi Steven, thanks for your help. I trigger the call throw .call file -- Attempting call on Local/[EMAIL PROTECTED]/n for [EMAIL PROTECTED]:1 (Retry 2) -- Executing SetAccount(Local/[EMAIL PROTECTED],2, 217815) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, IAX2/[EMAIL PROTECTED]/541141091828) in new stack -- Called [EMAIL PROTECTED]/541141091828 -- Call accepted by 217.14.132.162 (format G729A) -- Format for call is G729A -- IAX2/voiptalk/2 is making progress passing it to Local/[EMAIL PROTECTED],2 -- IAX2/voiptalk/2 answered Local/[EMAIL PROTECTED],2 Okay, now lets see your .call file. Still digging in till we see the dial that initiates the call. -- Steven Critchfield [EMAIL PROTECTED] .call file: Channel: Local/[EMAIL PROTECTED]/n MaxRetries: 2 RetryTime: 60 WaitTime: 20 Context: Validate Extension: 9545 Priority: 1 Account: 217815 ___ 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] Linux OS platforms
I have a question regarding to OS platform. As I see on Wiki -s homepage there are many type of linux version.And in some of them there are reported errors (regarding to asterisk ) for exemole in rad hat . Can you tell me what is the best linux paltform ,( version ), which is supported by digiroom card (T1 and TMD )and asterisk run on it stable ?. Which linux is prefereable ? for asterisk ? Hanks , Roby. Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'___ 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] Re: Voicemail not working properly
at first it was not answering (there was complete silence after 200 Ok and ACK). i dont know what was the reason. but now it is answering me(asking for mailbox then password). but the problem that is is not authenticating me to check mailbox i have defined mailbox and 1234 password (it is saying that invalid mailbox or password). i want to know what should be the datatype for mailbox and password. when i try to check my mailbox it is giving me error (invalid username or password). can i use varchar insted of int for mailbox and password. CREATE TABLE voicemail_table ( uniqueid int(11) NOT NULL auto_increment, customer_id int(11) NOT NULL default '0', context varchar(50) NOT NULL default '', mailbox varchar(5) NOT NULL default '0', password varchar(4) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '', pager varchar(50) NOT NULL default '', options varchar(100) NOT NULL default '', stamp timestamp(14) NOT NULL, PRIMARY KEY (uniqueid), KEY mailbox_context (mailbox,context) ) TYPE=MyISAM; __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
-Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] On February 8, 2005 08:44 am, David Brodbeck wrote: The sequence I hear on the extension, when I plug in an analog phone, is the click of the phone at the other end being hung up, followed immediately by a * touchtone. Then there's silence until I hang up. Hmm... I bet it has everything to do with not having 't' or 'T' in the dialplan -- asterisk is ignoring the tones because it's potentially a security problem. I have a t in the dialplan, but not T. I could add a T entry and see if it'll help. Currently what happens is Asterisk doesn't seem to notice the *, then it eventually goes to the t extension. This is undesirable since t transfers to the receptionist, who then gets a dead call. What puzzles me is it works fine if I dial *, but if I hang up instead and the PBX sends *, Asterisk doesn't seem to get it. At least that's my current working theory -- I am not sure if t/T listen ofr all DTMF or just #, and I also don't know the direction of your call (PBX - * or * - PBX). PBX - Asterisk. I'm setting up Asterisk to replace an old voice mail system. ___ 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] warning message
-Original Message- Good day all.I get the warning message on my system,this is for a snom 220,it repeats this message a few times,please help Feb 8 09:29:26 WARNING[1093445952]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 105 (Non-critical Request) Is there a page that describes all asterisk's error and warning messages? Thanks Altus /SNIP/ This message typically represents a NAT Issue, where in the SIP Client(SNOM 220) and Asterisk(Server) are not able to recognize each other's IPs to transmit packets successfully during the initial handshake. Using a STUN Server in the SNOM configuration would solve the problem and establish the call. I guess you are using DHCP on your network and the SNOM gets the IP from the Router in the Local Address ranges like 192.168.1.X or some such NAT IP. This Address being not a Public IP Address, you need to enable Network Address Translation with a Port Mapping for your Local IP Seshu Kanuri NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] DASS II cards supported
I know Q931 cards are supported, does anybody know how to go about supporting DASS II ? Thanks Stephen ___ 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] SER Interaction: Agents and Extensions
Hey gang, I'm trying to work out all possible scenarios using SER Asterisk in our upcomming deployment. The example scenario is 50 different customers, all with different numbers of SIP UAs. All UAs would register with SER; This will help keep any inter-office conversations off our bandwidth since SER doesn't handle the RTP stream. Calls from PSTN to UA are easy to handle. Asterisk receives the call on Zap card, and forwards to SER. SER looks up in its alias table which UA to send it to, and sends it. Calls from UA to PSTN are even easier. However, billing comes into question here. If every SIP call comes in to Asterisk from SER, how can I differentiate one customer from another? AFAIK, SER has no notion of 'context'. So, if offices wanted 4 digit extensions, I would be unable to duplicate any extensions right? Onto Agents; Say I setup *80 as the AgentLogin/Logoff number for a paticular customer. SER passes it on to Asterisk. I'd still have to setup contexts and agents just like normal right? With all of these caveats, it seems to me that a SER-Asterisk solution isn't that great. If anyone else out there can show me otherwise... Thanks, Matthew ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
On February 8, 2005 09:28 am, David Brodbeck wrote: What puzzles me is it works fine if I dial *, but if I hang up instead and the PBX sends *, Asterisk doesn't seem to get it. With you listening in on the same physical 2-wire that the PBX uses and you send *, does Asterisk see it? If you're on a call from the PBX to Asterisk and dial * from the PBX phone, does * see it? -A. ___ 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] Linux OS platforms
Which linux is prefereable ? for asterisk ? As long as you know how to rebuild your kernel, how to install modules, and how to manage basic unix security, the best Linux for Asterisk is the one you're most comfortable with. ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
-Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] On February 8, 2005 09:28 am, David Brodbeck wrote: What puzzles me is it works fine if I dial *, but if I hang up instead and the PBX sends *, Asterisk doesn't seem to get it. With you listening in on the same physical 2-wire that the PBX uses and you send *, does Asterisk see it? If you're on a call from the PBX to Asterisk and dial * from the PBX phone, does * see it? Yes, in both cases. ___ 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] codec order, does it matter
Does the order in which you allow codecs matter? cuz i've found that somethings work better if you allow them in a particular order. Alot of warnings and errors can also be eliminated. ___ 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] high-quality, high-bandwidth codecs?
On Tue, Feb 08, 2005 at 02:58:01PM +0100, Roy Sigurd Karlsbakk wrote: are there any codecs around that allows high quality as in studio lite? it may consume high bandwidth, and hopefully allow some packet loss. I'm not sure what studio lite means to you. Maybe hard figures would be more precise. G.722 might be interesting : 64 kbps, 7 kHz. It's not free. Otherwise, MP3 or OGG might be ok ? Would it be hard to do a codec_ogg? It would rather be a codec_vorbis, as Steve pointed out. It's definetly feasible. However, I'm not sure how useful it would be. You'd need some kind of device talking Vorbis to Asterisk. Does it exist ? -- Nicolas Bougues ___ 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] VoIP extn number planning
On Tue, 2005-02-08 at 06:27 -0600, Rich Adamson wrote: Looking for some advanced thoughts relative to exten number assignments. We're in the planning stage for rolling out asterisk at multiple small US telco/isp operations. Their typical voip customer has had their pstn line for a long time and wants to keep the pstn line and number, but add voip to their existing home/soho arrangement. The approach that I have taken is... 1 - at each place that I have asterisk, register the users full number with e164.org (or equivalent) 2 - Make sure I do e.164 lookups as part of the normal process of placing a call... 3 - If a call comes in via VoIP - alter its CLID so it looks the same as an incoming telco call - which makes identifying and returning the call simple. Effectively - I use the dialling plan from Telco. Each site retains its 'historical' number - which is probably the same as everyone has in their Rolodex/Diary/PDA (etc) - so there is no customer learning - or dialing funny access codes - etc If the call does not get through - my system simply uses the Telco line - as in the old way. If your client calls anyone else who implements the same rules - they'll get through on VoIP too... and if I take a phone book, look up your customer and call the number given - I'll use VoIP too... The only time that I do not do any number lookups is to 911 or operator specific numbers... which in South Africa tends to be '10XXX' format. This works fine for any Asterisk installation that has both traditional (= fixed connection to telco) and VoIP circuits. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
Okay, the problem appears to be that I'm tone deaf. ;) I finally thought to turn on debugging on the channel. The PBX is sending D, not *. The programmer of the previous voice mail system (whose configuration I was cribbing from) seems to have made the same mistake. ___ 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] In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
On February 8, 2005 09:48 am, David Brodbeck wrote: With you listening in on the same physical 2-wire that the PBX uses and you send *, does Asterisk see it? If you're on a call from the PBX to Asterisk and dial * from the PBX phone, does * see it? Yes, in both cases. How short is the * tone that the PBX is sending? You may want to actually use Monitor() to record the call and attach it to a bug, even just to see what the People In The Know have to say. -A. ___ 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] Using a Dual WAN Load Balancing Device
Pedro, My understanding is that this will not allow for any balancing on any connections once they are established. Any connection on the first line that is already established will continue to stay on that line/ip address until the connection is dropped and a new one is established. It would be better if you could get the ISP to set up the lines so that you could have a shadow DSL line like I've seen done with T1 lines before. Then this might be a more adequate solution in my mind. Jared Armstrong -Original Message- From: Pedro [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 08, 2005 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Using a Dual WAN Load Balancing Device We have a client that wants to bond 2 DSL circuits instead of getting a T-1 (or similar) at their office to run their VoIP traffic on. We came across this Multihomed Gateway (MH200): http://www.cyberpathinc.com/mh200/details.htm Does anybody think this would work if installed at the client location handling NAT for 10 Cisco 7960's and connecting to our public asterisk server? My concern (as is others on this list in regards to load balancing) is what would happen if a call had to be directed out the other WAN port of the MH200 or if a call were to come in on 1 circuit and it runs out of bandwidth - how would the call be delivered to the second circuit. Or even if during a call, the inbound audio is fine (since DSL usually has more bandwidth on the download), but the outbound audio stream had to be pushed out the other WAN port. Hope that all makes sense (I almost confused myself! LOL) I am not holding my breath that this is a viable solution, but was just wondering your thoughts. Thanks! Pedro ___ 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] how to pop up called number details using phpscripts in agi scripts
I got the called-name lookup going using php: http://muware.com/asterisk If you want to pop up additional details, you'll need a client application to notify a computer near the extension -- this is possible, but will require quite a bit more work. -Original Message- From: Mazhar Hussain [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 08, 2005 7:15 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how to pop up called number details using phpscripts in agi scripts Hi to all, I and using asterisk with following setup. 1. TDM400p card with four FXS modules, so there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 I want your guidance for the following issue. with help of agi scripts i am able to insert caller id number in database of mysql now i want to pop it up via html or php page but can any one of you let me know how can i use php scripts in agi scipts so that i can pick called id number from database and retreive the caller record from database using this called id number cheers mazhar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Autodetecting faxes
Checkout http://www.voip-info.org/wiki-NVBackgroundDetect I haven't had a chance to try it yet, but supposedly it works on SIP, ZAP, and IAX. On Tue, 8 Feb 2005 21:26:28 +1100, Mike Sander [EMAIL PROTECTED] wrote: That's all very well, but what do you do if you only have SIP extensions and IAX trunk - no Zaptel card. Will Fax detection still work at all? Thanks Mike ___ 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] ASTCC simultenous calls per card
Hi guys, do you know if it's possible to handle more than 1 call per card with astcc ? Thank you. ___ 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
How do I match a D? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
-Original Message- From: David Brodbeck [mailto:[EMAIL PROTECTED] Okay, the problem appears to be that I'm tone deaf. ;) I finally thought to turn on debugging on the channel. The PBX is sending D, not *. The programmer of the previous voice mail system (whose configuration I was cribbing from) seems to have made the same mistake. Is there some trick for matching the letter tones? I added this extension: exten = D,1,Goto(bye,s,1) But it doesn't trigger, even though I see this debugging output when I hang up: [ TYPE: DTMF (1) SUBCLASS: D (68) ] [Zap/1-1] ___ 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] Re: Using a Dual WAN Load Balancing Device
We have a client that wants to bond 2 DSL circuits instead of getting a T-1 (or similar) at their office to run their VoIP traffic on. We came across this Multihomed Gateway (MH200): http://www.cyberpathinc.com/mh200/details.htm Does anybody think this would work if installed at the client location handling NAT for 10 Cisco 7960's and connecting to our public asterisk server? My concern (as is others on this list in regards to load balancing) is what would happen if a call had to be directed out the other WAN port of the MH200 or if a call were to come in on 1 circuit and it runs out of bandwidth - how would the call be delivered to the second circuit. Or even if during a call, the inbound audio is fine (since DSL usually has more bandwidth on the download), but the outbound audio stream had to be pushed out the other WAN port. Hope that all makes sense (I almost confused myself! LOL) I am not holding my breath that this is a viable solution, but was just wondering your thoughts. I had the displeasure of working with the now defunct iSurfJanus from Amplify Networks which is similar to the MH200. I'm not sure the MH200 is capable of doing what you want it to do. I don't think it does incoming load balancing. The only ways I know of to host a machine behind two or more connections, incoming load balancing, are 1) BGP, 2) Cisco HSRP, or with 3) DNS and extremely short TTL values. There may be some other ways, but these are the popular ones. The multiple WAN devices capable of incoming load balancing like the F5 BigIP, Fatpipe Products, Radware Linkproof, etc. all use special DNS entries to spread the incoming connections between WAN connections. When I looked at the product specs of the MH200 it makes no mention of BGP, DNS, or anything else that might handle incoming connections. In fact, it doesn't say anything about incoming connections at all. To answer your question directly, I don't know how the other products work, but I could configure the iSurfJanus to respond to requests only on the same connection they came in on. If the MH200 does handle incoming connections, you will probably need to ask the folks that make it if you can explicitly specify to respond to incoming request on the same WAN connection they came in on. ___ 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: N Priority WAS Re: [Asterisk-Users] Voicemail timeouts after 30sec's everytime.
It's probably too late for me to say I don't want to sound like a jerk. :-P It was late and I get frustrated when people don't use available resources. I apologize. Anyways, a quick search of google.. http://www.google.com/search?q=asterisk%20n%20priority pulls up http://www.sineapps.com/news.php?rssid=160 On Tue, 8 Feb 2005 07:55:16 -0500, Michael George [EMAIL PROTECTED] wrote: I'm using stable rather than CVS. I did look on voip-info and I searched the mailing list archives. If there's another place I could've looked before asking, I'd love to know it to save redundant questions here in the future. Thanks! -- Is it something someone said, was it something someone said? ___ 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] agi command 'stream file' not working?
. Specifically, X is not a digit, you must either use for no interuptions permitted or use 0123456789 for all digits available to interupt. I also 'discovered' that you cannot send a sequence of commands to asterisk without reading the results between each command submission. Similar to the use of the manager interface. Thanks Marv Horst ___ 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] SRV lookups
Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for [EMAIL PROTECTED] the call is mapped to [EMAIL PROTECTED] Is that correct? If so, I have a problem: if somebody calls [EMAIL PROTECTED], Asterisk receives only the foo part. If someone calls [EMAIL PROTECTED], it receives john as the extension. Now the main question is: how do I know which SIP address the call originally went to? If I lose the domain name prone to SRV lookup, I can't decide where to route the call - for example, there may be [EMAIL PROTECTED] and [EMAIL PROTECTED], both get mapped to my central Asterisk server - I'm unable to know which of the john.s is being called, hence I cannot route the call correctly. Hope the question is clear enough ;) TIA, Robert -- Mit freundlichen Grüßen Robert Spielmann - TAL.DE Klaus Internet Service GmbH [EMAIL PROTECTED] Robertstr. 6 * D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 * Fax +49 (0) 202 / 495-399 ___ 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] Re: Using a Dual WAN Load Balancing Device
Noah, Thanks for your input on this. I am not sure if it handles incomng connections or not - will have to check. I don't think it will work either - worth a shot to ask though. Thanks! - Pedro On Tue, 8 Feb 2005 10:26:48 -0500, Noah Miller [EMAIL PROTECTED] wrote: We have a client that wants to bond 2 DSL circuits instead of getting a T-1 (or similar) at their office to run their VoIP traffic on. We came across this Multihomed Gateway (MH200): http://www.cyberpathinc.com/mh200/details.htm Does anybody think this would work if installed at the client location handling NAT for 10 Cisco 7960's and connecting to our public asterisk server? My concern (as is others on this list in regards to load balancing) is what would happen if a call had to be directed out the other WAN port of the MH200 or if a call were to come in on 1 circuit and it runs out of bandwidth - how would the call be delivered to the second circuit. Or even if during a call, the inbound audio is fine (since DSL usually has more bandwidth on the download), but the outbound audio stream had to be pushed out the other WAN port. Hope that all makes sense (I almost confused myself! LOL) I am not holding my breath that this is a viable solution, but was just wondering your thoughts. I had the displeasure of working with the now defunct iSurfJanus from Amplify Networks which is similar to the MH200. I'm not sure the MH200 is capable of doing what you want it to do. I don't think it does incoming load balancing. The only ways I know of to host a machine behind two or more connections, incoming load balancing, are 1) BGP, 2) Cisco HSRP, or with 3) DNS and extremely short TTL values. There may be some other ways, but these are the popular ones. The multiple WAN devices capable of incoming load balancing like the F5 BigIP, Fatpipe Products, Radware Linkproof, etc. all use special DNS entries to spread the incoming connections between WAN connections. When I looked at the product specs of the MH200 it makes no mention of BGP, DNS, or anything else that might handle incoming connections. In fact, it doesn't say anything about incoming connections at all. To answer your question directly, I don't know how the other products work, but I could configure the iSurfJanus to respond to requests only on the same connection they came in on. If the MH200 does handle incoming connections, you will probably need to ask the folks that make it if you can explicitly specify to respond to incoming request on the same WAN connection they came in on. ___ 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] ASTCC simultenous calls per card
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of thieumS Sent: Tuesday, February 08, 2005 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ASTCC simultenous calls per card Hi guys, do you know if it's possible to handle more than 1 call per card with astcc ? Astcc allows only a single simultaneous call per account number. Any pre-paid application will have some inherent problems in allowing multiple users/callers accessing the same pool of funds. Especially if the amount remaining in the account could be entirely used up by a single caller. Karl Putz Thank you. ___ 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] Broadvoice issues {Scanned}
I had problems as well. It was do to my sip.conf and extension.conf Here are my conf files. sip.conf [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to context=default ; Default context for incoming calls register = number:[EMAIL PROTECTED]/102 register = XX:[EMAIL PROTECTED] [broadvoice] ;-- This is what messed me up. This type=friend ; is up you will use in your exten username=XX ; line @broadvoice. fromuser=XX secret=sip-passwd host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes [bv-in] type=friend host=sip.broadvoice.com context=from-broadvoice dtmfmode=inband canreinvite=no nat=yes allow=ulaw extension.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [default] include = DISA exten = _0[1-9],1,Background,pls-hold-while-try exten = _0[1-9],2,Dial(SIP/[EMAIL PROTECTED]) exten = _1XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _911,1,Goto(911,911,1) exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _NXX,2,Dial(SIP/[EMAIL PROTECTED]) exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _8500,1,Goto(VoiceMail,8500,1) [from-broadvoice] exten = s,1,Wait(2) exten = s,2,Answer exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,5 exten = s,5,Background,welcome exten = s,6,Background,if-u-know-ext-dial exten = 8501,1,Goto(DISA,8501,1) exten = 8510,1,Goto(8510,8510,1) exten = i,1,Playback,pbx-invalid exten = i,2,Goto,s|6 exten = t,1,Goto,0|1 exten = _##,1,Hangup [from-pbx1] exten = 8510,1,Dial(SIP/8510,10) exten = 8510,2,Voicemail,u8510 [8510] exten = 8510,1,Wait(2) exten = 8510,2,Answer exten = 8510,3,Background,pls-hold-while-try exten = 8510,4,Dial(SIP/8510,10) exten = 8510,5,Background,pls-hold-while-try exten = 8510,6,Dial(SIP/[EMAIL PROTECTED],15) exten = 8510,7,Background,tt-somethingwrong exten = 8510,8,Voicemail,u8510 [VoiceMail] exten = 8500,1,VoicemailMain [DISA] exten = 8501,1,Answer exten = 8501,2,Wait,1 exten = 8501,3,DigitTimeout,5 exten = 8501,4,ResponseTimeout,10 exten = 8501,5,Authenticate(XXX) exten = 8501,6,DISA,no-password|default exten = i,7,Hangup [911] exten = _911,1,Background,no-911-1 exten = _911,2,Dial(SIP/8510,20) exten = _911,3,Goto(default,911,1) I hope this helps. David On Mon, 2005-02-07 at 14:36 -0800, Luki wrote: You are probably using your website password The password used for registering is the same you use for outgoing calls -- yes, it's different from your portal password. So if you can register and receive calls, you have the password you need. Double check that you use the section name from sip.conf in your dial plan, and that you have the correct password as well as the fromuser and username set in the broadvoice section in sip.conf. As Rich said before, post your relevant sip.conf (register statement and BV section) and your dialplan entry. --Luki ___ 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 -- David Shaw [EMAIL PROTECTED] ___ 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] AreskiCC Installation -- Please Help
Sounds like maybe you don't have either Postgres installed or PHP confirgured to use it. If you use RPMs, check for something in the php-pgsql family (%yum install php-pgsql) As a warning, you will also need to enable PHP globals in your php config. Hope that helps, J On Tue, 8 Feb 2005 09:17:54 -0500, shariq sajjad [EMAIL PROTECTED] wrote: Need Help .. I am trying to install AreskiCC Calling Card application but each time I tried to login as root -- I recieved this error Fatal error: Call to undefined function: pg_pconnect() in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 67 Please help me - I am stucked. I will appreciate your response. Thanks, Syed. ___ 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 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] faxing digium?
hi I've been trying to fax digium this agreement for a month or so now Any chance they can fix their fax? roy ___ 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] Re: Asterisk-Users Digest, Vol 7, Issue 113
Steve Blair writes I can redirect and relay calls to numerous destinations via SER but because the Octel needs an SMDI interface for mailbox identification I am stuck, none of the solutions thus far support SMDI-SIP munging. I just started thinking about the possibility of using Asterisk with a few FXS cards to provide the gateway between SIP and the Octel. The problem is I still need an SMDI channel that is integrated with the message processing part of the gateway. Has anyone look worked or developed any working models that might help? I have spent some time looking at this. There is no SMDI support for Asterisk but someone has posted a bounty for it. You might find that you can just replace the Octel with * but that's more of a jump than some are willing to take. There is the beginning of an APE derived C++ library for SMDI. ___ 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] Music on hold is a durge
I have just setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default = quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a standard 1.0.3 box, running headless (no x desktop) on FC2. On a P4 2.4GHz Any ideas? ___ 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] VoIP Termination in 479 Area Code
I am looking for termination of numbers in the 479 area code. I would like to either port them through my * box or direct SIP connection from the customer. I am in need of over 100 DID's. Anyone know of anyone that has this service besides Vonage or Packet8? --- Kelly D Griffin Network Engineer Tantella Wireless http://tantella.com 800.636.0306 Voice 479.464.8998 Fax ___ 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] SIP jitter?
how can I tune SIP jitter? is it possible today in asterisk? I assume you are asking for how to alleviate the effects of jitter on the RTP audio streams initated by SIP? Asterisk currently only has a jitter buffer for IAX, not for RTP streams. There are pland for the next generation jitter buffer code to hook into RTP as well. There is an entry on the bug tracker that touches on this topic. thanks is this in HEAD yet? roy ___ 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] CVS or release?
is the v1-0 CVS branch supposed to be stable as in STABLE, or should one use releases? v1-0 is the tag used for the latest changes to the stable branch. Releases are still your best bet, but if you are monitoring the CVS mailing list for commits to v1-0 stable, then you may see a patch go in that fixes some bug you've been having. The rule generally is don't run CVS unless you're monitoring the CVS mailing list. -rv1-0-5 will get you version 1.0.5 of Asterisk, which is the latest released version. is this static, as in same as the tar ball or will it change over time? roy ___ 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] Question about TDM11B Configuration
On Tue, 8 Feb 2005 11:56:18 +0200, Yousri Farouk [EMAIL PROTECTED] wrote: Hello all, i would like to configure TDM11B with Asterisk, if any one have the configuration steps please provide me it. Thanks in advance Have you tried looking at Digium's site?? http://www.digium.com/index.php?menu=documentation Try the wiki: voip-info.org ___ 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] VoIP Termination in 479 Area Code
Yes, We offer that stuff we can get numbers in most U.S area's Contact us 800-508-1251 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly Griffin Sent: Tuesday, February 08, 2005 11:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VoIP Termination in 479 Area Code I am looking for termination of numbers in the 479 area code. I would like to either port them through my * box or direct SIP connection from the customer. I am in need of over 100 DID's. Anyone know of anyone that has this service besides Vonage or Packet8? --- Kelly D Griffin Network Engineer Tantella Wireless http://tantella.com 800.636.0306 Voice 479.464.8998 Fax ___ 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 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] CVS or release?
The stable tree from cvs includes any patches since release that was also commited for the v1-0 tag since some issues were found after the release but not major enough for a new tar ball release. ___ 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
