Re: [asterisk-users] Probems receiving 200ok message
I already did this. I saw 200 ok arriving on internet side, but not in lan side. The correct was message 200 ok be forwarded to sip phone on lan side. Fred Em Sáb, 2007-12-08 às 18:39 +0200, Dovid B escreveu: Try running wireshark with a hub on the internet side and on the lan side and compare. - Original Message - From: Frederico Madeira To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 07, 2007 2:33 PM Subject: Re: [asterisk-users] Probems receiving 200ok message No. My asterisk server had two NIC, one for public internet and another to LAN for phones. The problem is when I receive SIP 200 from public internet. Thanks. Fred Em Qui, 2007-12-06 às 21:53 -0500, C F escreveu: is this machine or the phone behind nat? On 12/6/07, Frederico Madeira [EMAIL PROTECTED] wrote: Hi guys, Using tcpdump I could see the messages sip 200 arriving on my server, but enabling sip debug on asterisk console I only saw Invite and 180 message. What can be the source of this problem ? Thanks. Fred ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: Esta é uma parte de mensagem assinada digitalmente ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Probems receiving 200ok message
No. My asterisk server had two NIC, one for public internet and another to LAN for phones. The problem is when I receive SIP 200 from public internet. Thanks. Fred Em Qui, 2007-12-06 às 21:53 -0500, C F escreveu: is this machine or the phone behind nat? On 12/6/07, Frederico Madeira [EMAIL PROTECTED] wrote: Hi guys, Using tcpdump I could see the messages sip 200 arriving on my server, but enabling sip debug on asterisk console I only saw Invite and 180 message. What can be the source of this problem ? Thanks. Fred ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Probems receiving 200ok message
Hi guys, Using tcpdump I could see the messages sip 200 arriving on my server, but enabling sip debug on asterisk console I only saw Invite and 180 message. What can be the source of this problem ? Thanks. Fred ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No ACK on 200 OK
Hi guys, My asterisk didn't send ACK for 200 ok message just for one specific extension. The ATA used by this extension is used by other extensions, with same firmware version. Looking in wireshark, I saw that ATA sent 200ok and asterisk didn't confirm it with ACK. The ATA did this during 20s, after this, asterisk hangup the call. This issue happen only one asterisk start a call. I'm using the latest version of asterisk. What can be the source of this problem, and how can I debug this problem in asterisk ? Thanks. Fred ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Advanced Dial Plan
Hi Guys,
I Have this peers on my sip.conf
[provider-302333-3000]
type=friend
context=provider
secret=xpto
username=302000
host=sip.provider.com
fromuser=302000
insecure=very
canreinvite=no
[provider-30-3001]
type=friend
context=provider
secret=xpto
username=303001
host=sip.provider.com
fromuser=303001
insecure=very
canreinvite=no
I Have in my sip.conf two extension 3000 and 3001.
I have this rule in my extensions.conf
exten= _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten= _X.,2,Hangup
exten= _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten= _X.,2,Hangup
And every calls made by my both extension was using the first rule, so
calls from extension 3000 match with peer and work, but calls from
3001 didn't match with peer and I got error.
How can I use a conditional sentence like:
if {${CALLERID(num)}=3000)
{
exten= _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten= _X.,2,Hangup
}
else
if {${CALLERID(num)}=3001)
{
exten= _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten= _X.,2,Hangup
}
Thanks.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advanced Dial Plan
Philipp This didn't wotk. Let's suppose that my sip extension 3000 want to call to (302).123.3211 I need a rule in extensions.conf to match with this number, right ? So, I can't use rules that you advice. My problem is only for outbound calls. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/10/25, Philipp Kempgen [EMAIL PROTECTED]: Frederico Madeira wrote: I Have in my sip.conf two extension 3000 and 3001. I have this rule in my extensions.conf exten= _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt) exten= _X.,2,Hangup exten= _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt) exten= _X.,2,Hangup And every calls made by my both extension was using the first rule, so calls from extension 3000 match with peer and work, but calls from 3001 didn't match with peer and I got error. exten= 3000,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) exten= 3000,n,Hangup() exten= 3001,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) exten= 3001,n,Hangup() That dialplan is about as easy as it can get. :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advanced Dial Plan
Thanks for advices.
The last one from Tilghman fit better for my needs.
Thanks a lot.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
2007/10/25, Tilghman Lesher [EMAIL PROTECTED]:
On Thursday 25 October 2007 10:36:02 Brett Crapser wrote:
[outbound]
exten= _X.,1,GotoIf([${CALLERID(num)} == 3000]?path0|1)
exten= _X.,1,GotoIf([${CALLERID(num)} = 3000]?path0,${EXTEN},1)
exten= _X.,2,GotoIf([${CALLERID(num)} == 3001]?path1|1)
exten= _X.,2,GotoIf([${CALLERID(num)} = 3001]?path1,${EXTEN},1)
exten= _X.,3,Playback(tt-monkeys)
exten= _X.,4,Hangup
You do not need double '=' and you were missing the ${EXTEN}. Also, '|'
separating arguments is now deprecated.
--
Tilghman
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hide passwords in SIP.conf
Alex, In my sip.conf, I have some lines that i need to register on my voip provider and I have this configuration: register = user[:secret[:[EMAIL PROTECTED]:port][/extension] How can I hide or use md5 for this configuration ? Thanks. Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br Em Sex, 2007-10-19 às 16:47 -0400, Alex Balashov escreveu: Frederico, Take a look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret This is the only way I know of. -- Alex On Fri, 19 Oct 2007, Frederico Madeira wrote: Hi guys, There is other way instead plain text to define passwords in sip.conf ? In register, peers and extensions ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hide passwords in SIP.conf
Hi guys, There is other way instead plain text to define passwords in sip.conf ? In register, peers and extensions ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as Media Server
Hi Guys, I want to configure asterisk to act as media server on my network. I have one specific situation descibed bellow. Collect Calls 1. Subscriber A call to subscriber B 2. Gateway in A side, send this call to media server (asterisk) and the asterisk send the call to subscriber B 3. When B answer the call, MS should play prompt1 for B and prompt2 for A (prompts are differents but with same duration). 4, I will give a free time for this call during the time of prompt1 message How can I do this ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Loose routing method
Hi Guys, Where can I configure in asterisk if it should use strict routing or loose routing ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Enable Media Atribute on 180 Ringing
Hi guys, I've made some tests with a partner and when he call to me he can't hear ring back tone. My asterisk sent 180 ringing message to him. He told me that in 180 ringing there isn't a media attributes and i need to reconfigure my side to send 180 ringing with media attributes. How can i enable this on asterisk ? thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using gizmo as softphone for Linux
Hi guys, During a time, i'm looking for a softphone that work fine in linux, present good features, good audio quality and good interface. Recently i found gizmo and i'm testing it with my asterisk box. For my purpouse he fit perfectly. I'm worried about security, because i need to inform my asterisk username and password on gizmo interface, and i think that gizmo can store this information on a database, gizmo can recorder my conversation, can store my DTMF digits when i'm accessing my bank account using a IVR, etc... How can i minimize this issues ? It's possible to use gizmo stand alone of gizmo servers and only registered on my asterisk server ? Thanks. Fred signature.asc Description: Esta é uma parte de mensagem assinada digitalmente ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Two devices registrating same extension
Hi guys, Is it possible to asterisk manage multiple devices registration with same extension ? When asterisk receive a call for that extension, it send call to all devices registered with that extension, and rtp go to first one that answer the call. Thanks. Fred signature.asc Description: Esta é uma parte de mensagem assinada digitalmente ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems to transfer calls when it is ringing
Hi Guys, I've setup a asterik box on a trunk with alcatel 4200 pabx. When operator do a call for somedestination terminated by our asterisk he can't transfer this call until called party answer that call. He can't transfer call when it's only ringing. This is a issue of Asterisk or from Alcatel. This PABX have 2 ISDN links. One with asterisk and other with other carrier. Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Instability on Asterisk
Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Instability on Asterisk
Oquendo, My provieder require sip digest authebtication: Asterisk send register to sip provider sip provider response with 401 asterisk send register again with authentication header sip provider response ok This is normal process, when problem happen, this process ocour until 401 message, and asterisk didn't add auth header. I thins this is a problem in asterisk becouse my ip phones can't register into him. After few minutes asterisk can register again and ip phones too. Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, J. Oquendo [EMAIL PROTECTED]: Frederico Madeira wrote: Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. Why do you believe this to be an issue with Asterisk. What you describe is this in barebones YourNetworkPhones --- connect --- SIP Provider SIP Provider --- starts handshake --- Yournetwork YourNetwork --- gets ball rolling --- SIP Provider SIP Provider ... ignores you Sounds like you should be ripping into your SIP provider they're sending you unauthorized messages which sounds like either they changed something, or you did. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Instability on Asterisk
Matt, It's make no sense. Asterisk should process messages in diferents threds, not in queue. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, Matt [EMAIL PROTECTED]: While I don't use 1.4, it could be that the registration failure (you said 100 registration lines with your provider?!?) are blocking the phones from registering. This is only a guess, I don't know for sure. On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote: Oquendo, My provieder require sip digest authebtication: Asterisk send register to sip provider sip provider response with 401 asterisk send register again with authentication header sip provider response ok This is normal process, when problem happen, this process ocour until 401 message, and asterisk didn't add auth header. I thins this is a problem in asterisk becouse my ip phones can't register into him. After few minutes asterisk can register again and ip phones too. Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, J. Oquendo [EMAIL PROTECTED]: Frederico Madeira wrote: Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. Why do you believe this to be an issue with Asterisk. What you describe is this in barebones YourNetworkPhones --- connect --- SIP Provider SIP Provider --- starts handshake --- Yournetwork YourNetwork --- gets ball rolling --- SIP Provider SIP Provider ... ignores you Sounds like you should be ripping into your SIP provider they're sending you unauthorized messages which sounds like either they changed something, or you did. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Instability on Asterisk
I don't think that this problem is DNS, becouse asterisk can send register to my provider and he can replay to asterisk, so, DNS is working fine. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, Matt [EMAIL PROTECTED]: I understand... but I know, at least in 1.2 if there was a DNS failure for some reason asterisk stopped doing anything else. That is... if I restart asterisk and it goes to register with , say, my 6 SIP upstream peers... but they are timing out for some reason asterisk won't initialize zap, or other sip or IAX stuff until it times out all 6 of those. I was under the impression this was being fixed in 1.4, but maybe it has not been. On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote: Matt, It's make no sense. Asterisk should process messages in diferents threds, not in queue. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, Matt [EMAIL PROTECTED]: While I don't use 1.4, it could be that the registration failure (you said 100 registration lines with your provider?!?) are blocking the phones from registering. This is only a guess, I don't know for sure. On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote: Oquendo, My provieder require sip digest authebtication: Asterisk send register to sip provider sip provider response with 401 asterisk send register again with authentication header sip provider response ok This is normal process, when problem happen, this process ocour until 401 message, and asterisk didn't add auth header. I thins this is a problem in asterisk becouse my ip phones can't register into him. After few minutes asterisk can register again and ip phones too. Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, J. Oquendo [EMAIL PROTECTED]: Frederico Madeira wrote: Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. Why do you believe this to be an issue with Asterisk. What you describe is this in barebones YourNetworkPhones --- connect --- SIP Provider SIP Provider --- starts handshake --- Yournetwork YourNetwork --- gets ball rolling --- SIP Provider SIP Provider ... ignores you Sounds like you should be ripping into your SIP provider they're sending you unauthorized messages which sounds like either they changed something, or you did. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] Zap failure: cause 66 - Channel not implemented
Jose, Look for chan_zap.so into channel folder in your asterisk installation dir. If you only see chan_zap.c your module wasn't compiled. Try recompile asterisk. Run lsmod and see if zapata module is lodade, if not try modprobe zapata. In asterisk cli try to load module direct: module load chan_zap to see if any error happen. I had a similar problem using FC6 in kernel 2.6.20. To resolve I've use kernel i686 and kernel-devel i586. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/13, Jose Limeres [EMAIL PROTECTED]: Hi, I just compiled and installed Asterisk-1.4.2 along with zaptel-1.4.1 and libpri-1.4.0 on a Debian machine with a TDM400P card. Everything goes ok but when I try to make a call through the ZAP channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and zttool show the card correctly installed. When I tried to use the debug command ZAP SHOW, it was not present in the CLI. My question is: Is this an indication that Asterisk has not been correctly complied or is it normal that ZAP SHOW commands are not available in 1.4.2? Thanks for any hint. Jose Limeres -- Executing [EMAIL PROTECTED]:1] Dial(SIP/502-081d6f28, Zap/g1|60|to) in new stack [Apr 13 08:34:03] WARNING[4724]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 13 08:34:03] WARNING[4724]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/502-081d6f28' status is 'CHANUNAVAIL' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missing chan_zap.so
If your chan_zap module was compiled, try to load zaptel modules using: modprobe zaptel. In my case, i received an error and i need to use kernel in i686 and kernel-devel in 586 and recompile zaptel modules. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/11, Sanjay Rajdev [EMAIL PROTECTED]: Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. All SIP accounts were working fine, today I tried to install a fxs Sangoma A200 card and got the following error. [Apr 12 01:15:17] WARNING[31018]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 12 01:15:17] WARNING[31018]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) Searched google and came to conclusion that I was missing chan_zap.so on my machine. Followed the instruction of the bug at http://bugzilla.atrpms.net/show_bug.cgi?id=1165 and downloaded zaptel 1.4.1, after that executed the following commands ./configure make clean make make install Went to asterisk folder ./configure make clean make make upgrade But could not get chan_zap.so then did the make install of asterisk. still missing the chan_zap.so Can someone please help. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay to start sip registration after asterisk restart
Hi, My asterisk was working fine but today my calls won't out of my asterisk box. Restarting asterisk i need to wait around 10 min to can run sip show registry command. If i try to run this command before, i receive a error like: no such command. Why this happen ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What means: Request to schedule in the past?!?!
Hi guys, My asterisk is show me some errors on line registration. This message appear on console: Request to schedule in the past?!?! What it mean ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem on Asterisk to Register lines for out/in calls
Hi guys, I have a customer with asterisk registering 100 lines from my Voip Provider. In some times during a day we receive this messages: [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] NOTICE[26223]: chan_sip.c:7085 sip_reg_timeout: -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #3) [Feb 21 17:09:34] NOTICE[26223]: chan_sip.c:7085 sip_reg_timeout: -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #3) This message happen in all 100 lines. After few minutes all lines can register in my softswitch. This problem is not cyclical. Sniffing the network, i saw that asterisk send a register message, receive a 407 message and after instead send another register with authentication header, send another register message without authentication header. In most part of the time this asterisk work fine, except for this problems that happen 4 or 5 times per day. What could be the cause of this problem ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange problem
Hi guys. I'm working on a VOIP service provider. We have two customers running asterisk. Customer A and B. When A call to B everything is ok. When B call to A the call ring but sip messages didn't arrive on asterisk A. In my softswitch i see the invite sip message sended to A. When every other numbers(TDM and SIP) call do A everything is ok. Have any issue in asterisk that can resolve this problem ? I'm figuring out with our link provider to see if he has some firewall rules that can cause this problem I'm very very confuse becouse the invite message in every time come from my softswitch with ip of my softswitch so, why only invite originate on B side has this problem ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ERROR[2453]: chan_zap.c:8142 zt_pri_error: !! Unexpected Channel selection 3
Hi guys, I did an upgrade on one asterisk from 1.2.14 to 1.4.0, after this, all calls originated from PBX trunked with asterisk through TE110 board i receive this message: [Jan 16 21:19:42] ERROR[2453]: chan_zap.c:8142 zt_pri_error: !! Unexpected Channel selection 3 the call was completed and two ends talks normaly, the only incovenient is that message. Anybody know why this message appear ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to detect long calls
Hi guys, Look my example: pabx*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 64.71.xx.xx322121226ee03b46000 00103/15992 unkn No (d) Rx: BYE 64.71.xx.xx0113941735 57344d766af 00103/0 unkn No Tx: INVITE 64.71.xx.xx0113677599 5456e05e17d 00103/0 unkn No Tx: INVITE 64.71.xx.xx0113388754 3fe71d9114a 00103/0 unkn No Tx: INVITE 64.71.xx.xx0113388754 75c54f392c3 00103/0 unkn No Tx: INVITE 64.71.xx.xx0113677599 22fe2ae1237 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823241639 3b99e044545 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823231223 4345657f406 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823327211 5516645b4b7 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823336651 5692acca779 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823235526 14b7d28729f 00103/0 unkn No Tx: INVITE 64.71.xx.xx0793246319 3fe706487f1 00103/0 unkn No Tx: INVITE 64.71.xx.xx0613364414 13ea2109500 00103/0 unkn No Tx: INVITE 64.71.xx.xx0613364414 531f94b42c4 00103/0 unkn No Tx: INVITE 14 active SIP channels I can confirm that when i run this command, no one was in the office. What is this status ? Where can i see duration of this calls ? How can i kill them ? Thanks. Fred Em Ter, 2007-01-16 às 11:08 -0600, Savoy, Kevin - Williston, ND escreveu: We have been running an Asterisk box with 1.2.9.1 on it since August in a call center environment. We use the Asterisk box as an IVR and then pass the calls on to a Nortel Option 11C. Today we found in our long distance bill two calls that lasted a VERY long time. One was 58 hours and another was 38 DAYS!!! Nortel does not show this call being that long. Obviously the person that called in didn’t hold the line for 58 days so somehow between Asterisk and MCI the call got stuck open and didn’t hang up on the network. My question is two parts, part one, has anyone heard of anything like this where a call doesn’t hang up properly and seems “stuck” in the system. Part two is there anyway to monitor in Asterisk the length of all active calls and then if a call lasts longer then, say one hour, we could send off a text message or warning. Any ideas or comments would be helpful Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users image001.gif Description: image001.gif signature.asc Description: Esta é uma parte de mensagem assinada digitalmente ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Error on answer a SIP 401 message
Hi folks, Anyone already seen any problem like this ? Thanks. Fred Em Qua, 2007-01-03 às 16:24 -0300, Frederico Madeira escreveu: Hi, I'm a voip service provider and i'm setting up a asterisk box to register around 100 lines from my central softswitch. This asterisk box will be placed inside a customer and has a digium card to be interconected with customer's pabx. My problem is that when asterisk send register message, my softswitch return with sip 401 and asterisk should send a register message with Authorization in header. Only after 10, or 15 or 30 or 43 or 2 messages 401 asterisk start to send Authorization in header. This is a random time, don't follow any rule. This problem cause lines disregistration some times during a day. How can i solve this problem ? I use this parameters to register an account: register=number:[EMAIL PROTECTED]/number [fonar-number] type=friend context=default secret=pass username=number host=sip.provider.com fromuser=number fromdomain=sip.provider.com ;nat=yes ;insecure=very canreinvite=no ;qualify=1 dtmfmode=rfc2833 Thanks. signature.asc Description: Esta é uma parte de mensagem assinada digitalmente ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error on answer a SIP 401 message
Hi, I'm a voip service provider and i'm setting up a asterisk box to register around 100 lines from my central softswitch. This asterisk box will be placed inside a customer and has a digium card to be interconected with customer's pabx. My problem is that when asterisk send register message, my softswitch return with sip 401 and asterisk should send a register message with Authorization in header. Only after 10, or 15 or 30 or 43 or 2 messages 401 asterisk start to send Authorization in header. This is a random time, don't follow any rule. This problem cause lines disregistration some times during a day. How can i solve this problem ? I use this parameters to register an account: register=number:[EMAIL PROTECTED]/number [fonar-number] type=friend context=default secret=pass username=number host=sip.provider.com fromuser=number fromdomain=sip.provider.com ;nat=yes ;insecure=very canreinvite=no ;qualify=1 dtmfmode=rfc2833 Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Save SIP DEBUG output to a file
Hi guys, How can i save sip debug command output to a file ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disregistering Constantly - message: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181000' is now UNREACHABLE! Last qualify: 0
Hi guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181000' is now UNREACHABLE! Last qualify: 0 Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181001' is now UNREACHABLE! Last qualify: 0 Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer 'provider-13052181002' is now UNREACHABLE! Last qualify: 0 In all 100 numbers. I already change the link, but the problem still happpen. I use in sip.conf have this configuration to register lines on provider: register=13052181000:[EMAIL PROTECTED]/13052181000 register=13052181001:[EMAIL PROTECTED]/13052181001 register=13052181002:[EMAIL PROTECTED]/13052181002 . . . [provider-13052181000] type=friend context=default secret=1221212 username=13052181000 host=sip.provider.com fromuser=13052181000 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-13052181001] type=friend context=default secret=1221212 username=13052181001 host=sip.provider.com fromuser=13052181001 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-13052181002] type=friend context=default secret=1221212 username=13052181002 host=sip.provider.com fromuser=13052181002 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes If i disable 30 lines and restartr asterisk all lines are register normaly. So, Have any limit in network stack or in asterisk ? Have any tunning that can i make on linux or in asterisk to resolve this question ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls don't arrive for correct number
I have an asterisk box registering 100 numbers on a voip provider. Numers are: 2546.1000 to 2546.1099 My problem is that every incoming call arrived to number 2546.1099 that is the last number to register on voip provider. The correct is call arrive in destination number. See this exaple: I call to 2546.1000. -- Executing Dial(SIP/25461099-08738060, Zap/g1/3000) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/3000 -- Zap/1-1 is proceeding passing it to SIP/25461099-08738060 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/25461099-08738060 -- Hungup 'Zap/1-1' == Spawn extension (default, 25461000, 1) exited non-zero on 'SIP/25461099-08738060' How i solve this problem ?? See parts of my sip.conf register=25461000:[EMAIL PROTECTED]/25461000 register= 25461001:[EMAIL PROTECTED]/25461001 register=25461002:[EMAIL PROTECTED]/25461002 register= 25461003:[EMAIL PROTECTED]/25461003 . . . register=25461099:[EMAIL PROTECTED]/25461099 [provider-25461000] type=friend context=default secret= username=25461000 host=sip.provider.com fromuser=25461000 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-25461001] type=friend context=default secret= username=25461001 host=sip.provider.com fromuser=25461001 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-25461002] type=friend context=default secret= username=25461002 host=sip.provider.com fromuser=25461002 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes . . . [provider-25461099] type=friend context=default secret= username=25461099 host=sip.provider.com fromuser=25461099 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trunk Alcatel - Ring problem and call disconnection
Hi guys, Recentlly i did a asterisk gateway and use it with an alcatel pabx. All is working, i have only two problems. 1. When call incomming to asterisk, it forward to digium card to PABX Alcatel. The user that start the call can't hear the control tone of ring ring ring. Tha calls stay without sound until the called part answer the call. At this point, conversation follow normaly. 2. When an alcatel extension use asterisk to make a call, after some time, around 2 minutes the calls is hangup. How i can resolve this two problems ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk server reports
Hi guys, It's possible i scheduler in cron some kind of script or application that read asterisk logs and send via e-mail a complete report for pbx activity in specified period ?? I like to see how simultanios calls was made, total time in conversation, averege time of calls, most routes calls, etc Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit
Hi guys,I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P.Input callsVOIP Proider --- Asterisk --- Alcatel
Output CallsVOIP Proider --- Asterisk --- AlcatelIn alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems:1. When users dial 2 on phone (alcatel) they don't received a dial tone, only receive a ocuped tone;
2. When users make step one, in asterisk console i received this message:!! Unexpected Channel selection 3
-- Extension '' in context 'default' from '027' does not exist. Rejecting call on channel 0/31, span 1If i configure in alcatel short dialing such: if user dial 3020 alcatel sent do asterisk a block number 31122332. In this case works fine.
How i can solve this problem ??Bellow i list my extension.conf[default]ignorepat=0
### Internal Calls## Input Calls exten= 312120XX,1,Dial(Zap/g1/${EXTEN:-4})exten= 312120XX,2,Hangup### External Callsexten= _,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) # Local calls
exten= _,2,Hangupexten= _0XX,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) # Long distance Callsexten= _0XX,2Hangupexten= _00XX,1,Dial(SIP/[EMAIL PROTECTED]
,60,Tt) # Internacional Calls
exten= _00XX,2HangupThanks.-- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit
Thanks Leonardo,After change that parameter resolve the problem.Thans a lot.-- Frederico Madeira[EMAIL PROTECTED] www.madeira.eng.br 2006/11/9, Leonardo Gomes Figueira [EMAIL PROTECTED]: Frederico,Frederico Madeira escreveu: 1. When users dial 2 on phone (alcatel) they don't received a dial tone, only receive a ocuped tone; 2. When users make step one, in asterisk console i received this message: !! Unexpected Channel selection 3 -- Extension '' in context 'default' from '027' does not exist. Rejecting call on channel 0/31, span 1 If i configure in alcatel short dialing such: if user dial 3020 alcatel sent doasterisk a block number 31122332. In this case works fine. How i can solve this problem ??On zapata.conf:overlapdial=yesLeonardo___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with register command in SIP.conf
I'm registering 5 lines on my asterisk box from one voip provider.Lines;4040.4040.00014040.00024040.00034040.0004All lines is registered in 5060 port so when someone call to 4040.0001 the call arrive on asterisk but arrive to last number registered 4040.0004 becouse it is listening on same port as all others.How i make each number register in one different port, like 5060,5061,5062,5063 and 5064 ?? Thanks.-- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Direct call vs Block call
Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. For alcatel users use asterisk lines, should dial 0 to take tone from asterisk. In default configuration in alcatel, if user dial 0 this error occour: !! Unexpected Channel selection 3 -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/31, span 1 In alcatel we're enable block dial, so alcatel only send to asterisk when user end dialing all digits. How i permit the first case to work ?? Thanks. -- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF detection problem in PABX trunk
Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection problem in PABX trunk
I'm already try this configuration, but don't have sucess.-- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br 2006/10/27, Al Bochter [EMAIL PROTECTED]: Check your dtmfmode I use dtmfmode=rfc2833 Check with your provider Best regards,Al BochterBochter Services(Voip PBX) Toll Free: 866-638-1254 EXT: 250(Voip PBX) Free World DialUp: 780217 EXT: 250(Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Frederico Madeira wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0643-6, 10/26/2006 - 10/27/2006 7:52:49 AM ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] !! No channel map, no channel, and no ds1? What am I supposed to identify?
Hi guys,
I'm tryng to estabilish a trunk with an Alcatel 4200 Pabx. In asterisk i'm using a Digium TE110P as E1.
The span is ok with green led, but when pabx make calls to asterisk, i received this error:
asterisk*CLI
!! Unexpected Channel selection 3
-- Accepting call from '3069' to '30818559' on channel 1/31, span 1
-- Executing Dial(Zap/31-1, SIP/[EMAIL PROTECTED]|20|Tt) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/fp-33133000-09fdfa90 is ringing
!! Unexpected Channel selection 3
-- SIP/fp-33133000-09fdfa90 answered Zap/31-1
!! No channel map, no channel, and no ds1? What am I supposed to identify?
!! Unable to add IE 'Channel Identification'
== Spawn extension (default, 30818559, 1) exited non-zero on 'Zap/31-1'
-- Hungup 'Zap/31-1'
Sep 23 20:13:25 WARNING[3765]: chan_zap.c:8788 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1
Sep 23 20:13:29 WARNING[3765]: chan_zap.c:8788 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1
My configuration files is:
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
trunkgroup = 1,16
spanmap = 1,1,1
language=uk
context=default
switchtype=euroisdn
signalling=pri_net
group=1
callgroup=1
pickupgroup=1
immediate=no
echocancel=yes
channel = 1-15,17-31
/etc/asterisk/extensions.conf
# SIP - Alcatel
exten= 331330XX,1,Dial(Zap/g1/${EXTEN})
exten= 331330XX,2,Hangup
# Alcatel - SIP
exten= _,1,Dial(SIP/[EMAIL PROTECTED],20,Tt) #
exten= _,2,Hangup
What can be hrong in this configuration ???
Thanks.
-- Frederico Madeira[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - alcatel
I'm trying the same in Alcatel 4200 and i solved changing ignaling from pri_cpe for pri_net.-- Frederico Madeira[EMAIL PROTECTED] 2006/9/26, Sylvain ZUCCA [EMAIL PROTECTED]: Hi, can you send logs from alcatel 4400 ? just log in with account mtcl and launch t3 to see traces from the PBX Best Regards. 2006/9/26, et pourquoi pas ? epp [EMAIL PROTECTED]: Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? Best regards, Thomas ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - alcatel
Nicolas, We use a TE110P from digium. We make the same procedures oriented in that website. the only change was in signaling as i've said previously. My alcatel aready have an E1 ISDN installed from local carrier. After asterisk is setup, we change cables from carrier to asterisk, and our span stay in green state. Wich pins of cable you use in ISDN cable ?? What is the result of zttools -v ??? After span configuration we have problemas making calls, se my post in other forum: http://forums.digium.com/viewtopic.php?t=9868highlight=alcatel+4200 -- - Frederico Madeira [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] ICQ: 37152149 SKYPE: fred_madeira Registered GNU/Linux n 206120 -- Powered by LINUX -- CCNA Em Ter, 2006-09-26 s 23:25 +0200, Nicolas Bocquet escreveu: Hello, We have test this configuration but we think it's a problem with the Alcatel. how are you doing to make the trunk between alcatel and Asterisk? We use a card PRA recommended by an Alcatel's technician and you? Thanks Nicolas On 9/26/06, Frederico Madeira [EMAIL PROTECTED] wrote: I'm trying the same in Alcatel 4200 and i solved changing ignaling from pri_cpe for pri_net. -- Frederico Madeira [EMAIL PROTECTED] 2006/9/26, Sylvain ZUCCA [EMAIL PROTECTED]: Hi, can you send logs from alcatel 4400 ? just log in with account mtcl and launch t3 to see traces from the PBX Best Regards. 2006/9/26, et pourquoi pas ? epp [EMAIL PROTECTED]: Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? Best regards, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: Esta é uma parte de mensagem assinada digitalmente ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Trunk with Alcatel 4200 PABX
Hi guys,
I'm tryng to estabilish a trunk with an Alcatel 4200 Pabx. In asterisk i'm using a Digium TE110P as E1.
The span is ok with green led, but when pabx make calls to asterisk, i received this error:
asterisk*CLI
!! Unexpected Channel selection 3
-- Accepting call from '3069' to '30818559' on channel 1/31, span 1
-- Executing Dial(Zap/31-1, SIP/[EMAIL PROTECTED]|20|Tt) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/fp-33133000-09fdfa90 is ringing
!! Unexpected Channel selection 3
-- SIP/fp-33133000-09fdfa90 answered Zap/31-1
!! No channel map, no channel, and no ds1? What am I supposed to identify?
!! Unable to add IE 'Channel Identification'
== Spawn extension (default, 30818559, 1) exited non-zero on 'Zap/31-1'
-- Hungup 'Zap/31-1'
Sep 23 20:13:25 WARNING[3765]: chan_zap.c:8788 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1
Sep 23 20:13:29 WARNING[3765]: chan_zap.c:8788 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1
My configuration files is:
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
trunkgroup = 1,16
spanmap = 1,1,1
language=uk
context=default
switchtype=euroisdn
signalling=pri_net
group=1
callgroup=1
pickupgroup=1
immediate=no
echocancel=yes
channel = 1-15,17-31
/etc/asterisk/extensions.conf
# SIP - Alcatel
exten= 331330XX,1,Dial(Zap/g1/${EXTEN})
exten= 331330XX,2,Hangup
# Alcatel - SIP
exten= _,1,Dial(SIP/[EMAIL PROTECTED],20,Tt) #
exten= _,2,Hangup
What can be hrong in this configuration ???
Thanks.
-- Frederico Madeira[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
