Re: [asterisk-users] Re: [asterisk-dev] G729 Replacement Codec - FREE or may ne cheaper than existing one.
On 9/2/06, Daniel Pocock [EMAIL PROTECTED] wrote: http://www.readytechnology.co.uk/open/ipp-codecs Asterisk 1.2 support coming shortly. Asterisk 1.2 support? I'm using your codecs ever since 1.2 was released? Even though Asterisk always complains about modules being present that may be incompatible with Asterisk 1.2 after doing make install everything seems to work fine? What will be new / different for the codecs with 1.2 support? (Maybe it would be even better to work on 1.4 support since that will be out soon and skip 1.2 as a whole?) ___ --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] SER+Asterisk integration
I have ser sitting on my iptables nat box and my asterisk box on the lan
. Ser does forwarding so that any requests (register,invite,ack,...) to
the nat box at 5060 r sent to my asterisk box on the lan .I can register
from outside
to my asterisk box but there is only one way audio , reason being that
when the asterisk box sends a sip packet whith session description the sdp
part of the sip packet is not natted .I have tried the following :
if(src_ip==10.0.0.0/255.0.0.0){
force_rtp_proxy();
encode_contact(enc_prefix,wanip);
sdp_mangle_ip(10.0.0.0/255.0.0.0,wanip);
};
and it does not work because my ethernet dump shows that the contact in
sdp is
not mangled.
___
--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] File structure question
On Thu, Aug 31, 2006 at 03:52:00PM -0500, Jay Moore wrote: I have a question on how I can better organize my .conf files. I have 3 different groups of people who use my VoIP service. Let's call them 'Office', 'Factory' and 'Public'. In my Asterisk directory, I have created three folders: 'office', 'factory' and 'public', inside each of which has a sip.conf and an extensions.conf file with appropriate account and extension information. Say, for example, I need to limit some users of the 'Public' group so they cannot make calls outside the building. Obviously I would create two separate contexts. One for users who can make calls outside the build, and one for users who cannot. I would then assign the appropriate context to each user. Right now, I have each appropriate context defined in the main extensions.conf. What I'd like to do is reduce the clutter in extensions.conf and move each context into the extensions.conf in the appropriate subfolder. How do I tell the main extensions.conf file to include the other extensions.conf files without putting an #include file in a context of its own? I hope what I've explained makes sense. If not, please ask questions and I'll try to answer. #include is a verbatim text include. if extensions.conf has: [main] exten = aaa,1,Line1 #include otherfile.conf exten = aaa,2,Line2 and othererfile.conf has: exten = aaa,2,OtherLine1 [other] exten = aaa,1,OtherLine2 You'll eventually get: main: aaa: 1. Line1 2. OtherLine1 other: aaa: 1. OtherLine2 2. Line2 -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Queue timeout problems
-Ursprüngliche Nachricht- Von: Mr. Jones [mailto:[EMAIL PROTECTED] Gesendet: Sonntag, 3. September 2006 06:10 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Queue timeout problems Thanks Guido - I tried that and still have the same problem. The call never seems to leave the queue. Any other ideas? Hmm, to have a closer look on the problem, one could do the following Activate debugging, error and verbose logging in logger.conf by having a line like this: console = notice,warning,error,debug,verbose Open the cli and do a logger reload set verbose to 5 or even 255 Initiate a call to the queue and watch for errors/informations. Perhaps, define a context named test and put a really simple command in it. Something like this [test] exten = 120,1,Answer() exten = 120,2,Playback(some-sound-file) exten = 120,3,Hangup Change your queue to call this context in the second priority. Also have a closer look on your include commands in the dialplan... Normally an extensions reload on the cli should activate the changes to the dialplan, but with a restart now you should be save. Good luck Guido ___ --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 not sending RTP
Hi All, Here's a funny bit of a problem. I've got an asterisk server which appears not to be sending any RTP out of the system. Any ideas why such a weird issue would arise? I've tested this scenario via several termination gateways with SIP, and always there was no RTP in either directions. More then that, when running tcpdump, it appears as if asterisk isn't even sending any RTP to the outbound SIP gateway. This was seen on both 1.2.10 and 1.2.11 -- Kind Regards, Nir Simionovich Chief Technology Officer Atelis PLC ___ --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] Using Thunderbird (mail client) to call Contacts from Address Book
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ajmcello wrote: I'm looking for a way to dial my contacts using a SIP or VOIP gateway in Thunderbirds Addressbook. I can do this using Outlook with SIPTAPI, ASTAPI, and a couple of others, however, I have not found a way to do so in Thunderbird. Anybody have any ideas? http://www.voip-info.org/wiki/view/Asterisk+TAPI http://www.snapanumber.com/ - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE+qUPS6d5vy0jeVcRAqPAAJ9g52YlokYvWi6VX/p5jwG01lKcPwCfU6ip J+BT83H7D3KaCS7pg7zbq9I= =HfOa -END PGP SIGNATURE- ___ --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 not sending RTP
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nir Simionovich wrote: Hi All, Here's a funny bit of a problem. I've got an asterisk server which appears not to be sending any RTP out of the system. Any ideas why such a weird issue would arise? I've tested this scenario via several termination gateways with SIP, and always there was no RTP in either directions. More then that, when running tcpdump, it appears as if asterisk isn't even sending any RTP to the outbound SIP gateway. This was seen on both 1.2.10 and 1.2.11 Are you dumping just one ethernet card? Is it possible it's trying to send out the other? What do you have in the way of a firewall? Do you have iptables rules? Does the SIP protocol get sent? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE+qgqS6d5vy0jeVcRAkW5AJkByL6OQXWzwrlVLqHu1mlk1Ycp8ACfeTDw 3ZRsVgEl7lLYMZgyGCJ08Ak= =kd0/ -END PGP SIGNATURE- ___ --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] UK (BT) Problem with TDM 400P
I'm trying to get my TDM400P to work with a BT POT line. I've done everything I can think of to get the uk settings right (in zapata.conf, zaptel.conf and options for the wctdm driver) - and they all look right (ie uk like) and look like they are working when I try diagnostics with lspci, dsmsg, asterisk -c etc - but when I connect the FXO unit to the BT line it just makes it (the BT line) go permanently engaged. I'm tearing my hair out and about to chuck it all in the bin, but before I do, has anyone ever managed to get a TDM400P to work with a BT line and did they have any of these issues? Thanks for any help Mark ___ --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] UK (BT) Problem with TDM 400P
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mark Muffett wrote: I'm trying to get my TDM400P to work with a BT POT line. I've done everything I can think of to get the uk settings right (in zapata.conf, zaptel.conf and options for the wctdm driver) - and they all look right (ie uk like) and look like they are working when I try diagnostics with lspci, dsmsg, asterisk -c etc - but when I connect the FXO unit to the BT line it just makes it (the BT line) go permanently engaged. I'm tearing my hair out and about to chuck it all in the bin, but before I do, has anyone ever managed to get a TDM400P to work with a BT line and did they have any of these issues? You sure it's an FXO module? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE+q+WS6d5vy0jeVcRArUZAJ9a8+wU+MQSlrc+Vuk6XL45tbccFQCcCy3O ZdLMujY7KMcgC/bwBUfWlMo= =MBcZ -END PGP SIGNATURE- ___ --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] UK (BT) Problem with TDM 400P
On Sun, Sep 03, 2006 at 11:05:35AM +0100, Mark Muffett wrote: I'm trying to get my TDM400P to work with a BT POT line. I've done everything I can think of to get the uk settings right (in zapata.conf, zaptel.conf and options for the wctdm driver) - and they all look right (ie uk like) and look like they are working when I try diagnostics with lspci, dsmsg, asterisk -c etc - but when I connect the FXO unit to the BT line it just makes it (the BT line) go permanently engaged. I'm tearing my hair out and about to chuck it all in the bin, but before I do, has anyone ever managed to get a TDM400P to work with a BT line and did they have any of these issues? Please convince us that you have done that right. For starters: does 'zap show channels' show a channel ? Is the line OK? Does a normal phone work well? What is the output of 'cat /proc/zaptel/*' ? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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 not sending RTP
Hi Matt, I'm dumping only the eth0 interface, as this is the only interface configured on the box, eth1 is disabled. IPtables is completely disabled on the server, so that is not the issue. SIP invites appear to be handled correctly and being sent in and out correctly. Other than that, I'm fairly baffled. Nir s - Original Message - From: Matt Riddell (IT) [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, September 3, 2006 1:02:18 PM GMT-0800 Subject: Re: [asterisk-users] Asterisk not sending RTP -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nir Simionovich wrote: Hi All, Here's a funny bit of a problem. I've got an asterisk server which appears not to be sending any RTP out of the system. Any ideas why such a weird issue would arise? I've tested this scenario via several termination gateways with SIP, and always there was no RTP in either directions. More then that, when running tcpdump, it appears as if asterisk isn't even sending any RTP to the outbound SIP gateway. This was seen on both 1.2.10 and 1.2.11 Are you dumping just one ethernet card? Is it possible it's trying to send out the other? What do you have in the way of a firewall? Do you have iptables rules? Does the SIP protocol get sent? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE+qgqS6d5vy0jeVcRAkW5AJkByL6OQXWzwrlVLqHu1mlk1Ycp8ACfeTDw 3ZRsVgEl7lLYMZgyGCJ08Ak= =kd0/ -END PGP SIGNATURE- ___ --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 -- Kind Regards, Nir Simionovich Chief Technology Officer Atelis PLC ___ --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+ser+docs
Where can I find docs on ser and asterisk intergration. ___ --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] Unable to open pseudo channel for timing... Sound may be choppy.
On Sat, Sep 02, 2006 at 07:24:03AM +0200, Dominik Weber wrote: Hello, my name is dominik, and i'm using asterisk with voip without isdn, only sip. I've the following errors in my logfile: Unable to open pseudo channel for timing... Sound may be choppy. Cannot allow unknown format 'G711a' Unable to open IAX timing interface: No such file or directory But i'm not using IAX Can anyone help me ? You may need to install zaptel and the ztdummy (kernel) module as a timing source. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] UK (BT) Problem with TDM 400P
It's an FXO module (I've got two FXS modules on the same card and they work - I get a dialing tone from them), and the phone line works with an ordinary phone. dmesg gives: Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXO (UK mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules) Registered tone zone 4 (United Kingdom) asterisk -c -vvv starts up with no errors or warnings and zap show channels gives: *CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn 1from-internal 2from-internal 3from-pstn cat /proc/zaptel/* gives: asterisk1:/home/mark # cat /proc/zaptel/* Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 1 WCTDM/0/0 FXOKS 2 WCTDM/0/1 FXOKS 3 WCTDM/0/2 FXSKS 4 WCTDM/0/3 My /etc/zaptel is: asterisk1:/home/mark # cat /etc/zaptel.conf # Autogenerated by ./genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 fxoks=1 fxoks=2 fxsks=3 # channel 4, WCTDM, no module. # Global data loadzone= uk defaultzone = uk My zapata.conf is: asterisk1:/home/mark # cat /etc/zaptel.conf # Autogenerated by ./genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 fxoks=1 fxoks=2 fxsks=3 # channel 4, WCTDM, no module. # Global data loadzone= uk defaultzone = uk asterisk1:/home/mark # more /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [trunkgroups] ; ; Trunk groups are used for NFAS or GR-303 connections. ; ; [channels] ; context=test usecallerid=yes hidecallerid=no immediate=no #include zapata-channels.conf asterisk1:/home/mark # and my zaptel-channels.conf is: asterisk1:/home/mark # more /etc/asterisk/zapata-channels.conf ; Autogenerated by ./genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; ; Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 signalling=fxo_ks callerid=Channel 1 6001 mailbox=6001 group=5 context=from-internal channel = 1 signalling=fxo_ks callerid=Channel 2 6002 mailbox=6002 group=5 context=from-internal channel = 2 signalling=fxs_ks callerid=asrecieved mailbox= group=0 context=from-pstn cidsignalling=v23 cidstart=polarity ;;; line=3 WCTDM/0/2 FXSKS busydetect=no channel = 3 ; channel 4, WCTDM, no module. asterisk1:/home/mark # Any idea what more I can do before it drives me mad? Thanks for any help Mark On 03/09/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 03, 2006 at 11:05:35AM +0100, Mark Muffett wrote: I'm trying to get my TDM400P to work with a BT POT line. I've done everything I can think of to get the uk settings right (in zapata.conf, zaptel.conf and options for the wctdm driver) - and they all look right (ie uk like) and look like they are working when I try diagnostics with lspci, dsmsg, asterisk -c etc - but when I connect the FXO unit to the BT line it just makes it (the BT line) go permanently engaged. I'm tearing my hair out and about to chuck it all in the bin, but before I do, has anyone ever managed to get a TDM400P to work with a BT line and did they have any of these issues? Please convince us that you have done that right. For starters: does 'zap show channels' show a channel ? Is the line OK? Does a normal phone work well? What is the output of 'cat /proc/zaptel/*' ? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Asterisk not sending RTP
Title: RE: [asterisk-users] Asterisk not sending RTP I've now also enabled RTP debugging, and noticed that Asterisk doesn't send out RTP at all. All the lines appear as the following: Got RTP packet from 62.219.61.73:59436 (type 0, seq 1243, ts -1997588432, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1244, ts -1997588352, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1245, ts -1997588272, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1246, ts -1997588192, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1247, ts -1997588112, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1248, ts -1997588032, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1249, ts -1997587952, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1250, ts -1997587872, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1251, ts -1997587792, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1252, ts -1997587712, len 80) Got RTP packet from 62.219.61.73:59436 (type 0, seq 1253, ts -1997587632, len 80) Any ideas anyone ? Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Nir Simionovich Sent: Sunday, September 03, 2006 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk not sending RTP Hi Matt, I'm dumping only the eth0 interface, as this is the only interface configured on the box, eth1 is disabled. IPtables is completely disabled on the server, so that is not the issue. SIP invites appear to be handled correctly and being sent in and out correctly. Other than that, I'm fairly baffled. Nir s - Original Message - From: Matt Riddell (IT) [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, September 3, 2006 1:02:18 PM GMT-0800 Subject: Re: [asterisk-users] Asterisk not sending RTP -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nir Simionovich wrote: Hi All, Here's a funny bit of a problem. I've got an asterisk server which appears not to be sending any RTP out of the system. Any ideas why such a weird issue would arise? I've tested this scenario via several termination gateways with SIP, and always there was no RTP in either directions. More then that, when running tcpdump, it appears as if asterisk isn't even sending any RTP to the outbound SIP gateway. This was seen on both 1.2.10 and 1.2.11 Are you dumping just one ethernet card? Is it possible it's trying to send out the other? What do you have in the way of a firewall? Do you have iptables rules? Does the SIP protocol get sent? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE+qgqS6d5vy0jeVcRAkW5AJkByL6OQXWzwrlVLqHu1mlk1Ycp8ACfeTDw 3ZRsVgEl7lLYMZgyGCJ08Ak= =kd0/ -END PGP SIGNATURE- ___ --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 -- Kind Regards, Nir Simionovich Chief Technology Officer Atelis PLC ___ --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] Query about Call Detail Record in Asterisk
Hi, My testbed is as follows: sipphone -- Asterisk PBX 1 -- Asterisk PBX 2 -- PSTN -- Analog Phone I understand that one of the fields in the CDR (Call Detail Record) is the Answer field which is the time when call is answered. Is it right that : a) the Answer field of the CDR at Asterisk PBX 1 shows the time when Asterisk PBX 2 answers the call from Asterisk PBX 1 ? b) the answer field of the CDR at Asterisk PBX 2 shows the time when the Analog Phone answers the call from Asterisk PBX 2 ? regards, Kwang Mien ___ --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 not sending RTP
Nir Simionovich wrote: Any ideas anyone ? Do you have a compatible codec? What does the SDP show? Is sip.conf binding to a valid IP address? Jeremy McNamara ___ --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 not sending RTP
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nir Simionovich wrote: I've now also enabled RTP debugging, and noticed that Asterisk doesn't send out RTP at all. All the lines appear as the following: Got RTP packet from 62.219.61.73:59436 (type 0, seq 1243, ts -1997588432, len 80) What happens with a SIP Debug? Are you sure the session is set up properly? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE+uUUS6d5vy0jeVcRAr1KAJ9Og6hT1Tfi4ylLqFySfs/l4HR23wCfepmu a7ZeTntPJ+eXMDim7oO4M4M= =+IYG -END PGP SIGNATURE- ___ --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] UK (BT) Problem with TDM 400P
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mark Muffett wrote: It's an FXO module (I've got two FXS modules on the same card and they work - I get a dialing tone from them), and the phone line works with an ordinary phone. Have you tried not using the include? Does a normal phone attached to the line work? Does it work connected to an FXS socket (i.e. FXO - FXS)? Do you see anything when you ring the line? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE+uWrS6d5vy0jeVcRAog/AJwL6PikP+I0zaO6T4AXVRLxy8pj/ACfT0qk ly4/JGV1VyaLEsCsgdzm6cE= =TnVB -END PGP SIGNATURE- ___ --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] UK (BT) Problem with TDM 400P
Mark Muffett wrote: I'm trying to get my TDM400P to work with a BT POT line. I've done everything I can think of to get the uk settings right (in zapata.conf, zaptel.conf and options for the wctdm driver) - and they all look right (ie uk like) and look like they are working when I try diagnostics with lspci, dsmsg, asterisk -c etc - but when I connect the FXO unit to the BT line it just makes it (the BT line) go permanently engaged. I'm tearing my hair out and about to chuck it all in the bin, but before I do, has anyone ever managed to get a TDM400P to work with a BT line and did they have any of these issues? Thanks for any help Bad fxo module? Call digium support and let them help diagnose the problem. Also, the E/F revision is rather old; current is more like rev J. ___ --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 not sending RTP
Title: RE: [asterisk-users] Asterisk not sending RTP well, here is the full SIP debug: Sep 3 10:05:59 DEBUG[6139] manager.c: Manager received command 'Originate' Sep 3 10:05:59 DEBUG[6139] chan_sip.c: Setting NAT on RTP to 0 Sep 3 10:05:59 DEBUG[6139] chan_sip.c: Outgoing Call for 972544482826 Sep 3 10:05:59 VERBOSE[6139] logger.c: We're at 192.117.233.176 port 18372 Sep 3 10:05:59 VERBOSE[6139] logger.c: Adding codec 0x4 (ulaw) to SDP Sep 3 10:05:59 VERBOSE[6139] logger.c: Adding codec 0x8 (alaw) to SDP Sep 3 10:05:59 VERBOSE[6139] logger.c: 13 headers, 9 lines Sep 3 10:05:59 VERBOSE[6139] logger.c: Reliably Transmitting (no NAT) to 62.219.61.73:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.117.233.176:5060;branch=z9hG4bK33e91b50;rport From: 972544482826 sip:[EMAIL PROTECTED];tag=as5514479f To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 03 Sep 2006 09:05:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 188 v=0 o=root 6093 6093 IN IP4 192.117.233.176 s=session c=IN IP4 192.117.233.176 t=0 0 m=audio 18372 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Sep 3 10:05:59 VERBOSE[6124] logger.c: -- SIP read from 62.219.61.73:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport=5060 From: 972544482826 sip:[EMAIL PROTECTED];tag=as5514479f To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (6 headers 0 lines)Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (6 headers 0 li nes)--- Sep 3 10:05:59 DEBUG[6124] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a8dc38a2198b8fb0 [EMAIL PROTECTED]' Request 102: Found Sep 3 10:05:59 VERBOSE[6124] logger.c: -- SIP read from 62.219.61.73:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport=5060 From: 972544482826 sip:[EMAIL PROTECTED];tag=as5514479f To: sip:[EMAIL PROTECTED];tag=2607 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Supported: timer,100rel Content-Length: 0 Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (8 headers 0 lines)Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (8 headers 0 li nes)--- Sep 3 10:05:59 DEBUG[6124] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a8dc38a2198b8fb0 [EMAIL PROTECTED]' Request 102: Found Sep 3 10:06:00 VERBOSE[6124] logger.c: -- SIP read from 62.219.61.73:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport=5060 From: 972544482826 sip:[EMAIL PROTECTED];tag=as5514479f To: sip:[EMAIL PROTECTED];tag=2607 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Type: application/sdp Supported: timer,100rel Content-Length: 123 v=0 o=MG4000|2.0 14726 14726 IN IP4 62.219.61.73 s=- c=IN IP4 62.219.61.73 t=0 0 m=audio 51644 RTP/AVP 0 a=ptime:10 Sep 3 10:06:00 VERBOSE[6124] logger.c: --- (9 headers 7 lines)Sep 3 10:06:00 VERBOSE[6124] logger.c: --- (9 headers 7 li nes)--- Sep 3 10:06:00 DEBUG[6124] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a8dc38a2198b8fb0 [EMAIL PROTECTED]' Request 102: Found Sep 3 10:06:00 VERBOSE[6124] logger.c: Found RTP audio format 0 Sep 3 10:06:00 VERBOSE[6124] logger.c: Peer audio RTP is at port 62.219.61.73:51644 Sep 3 10:06:00 DEBUG[6124] chan_sip.c: Peer audio RTP is at port 62.219.61.73:51644 Sep 3 10:06:00 VERBOSE[6124] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), c ombined - 0x4 (ulaw) Sep 3 10:06:00 VERBOSE[6124] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Sep 3 10:06:01 VERBOSE[6124] logger.c: -- SIP read from 62.219.61.73:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport=5060 From: 972544482826 sip:[EMAIL PROTECTED];tag=as5514479f To: sip:[EMAIL PROTECTED];tag=2607 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Type: application/sdp Contact: sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone Supported: timer,100rel Content-Length: 123 v=0 o=MG4000|2.0 14726 14726 IN IP4 62.219.61.73 s=- c=IN IP4 62.219.61.73 t=0 0 m=audio 51644 RTP/AVP 0 a=ptime:10 Sep 3 10:06:01 VERBOSE[6124] logger.c: --- (10 headers 7 lines)Sep 3 10:06:01 VERBOSE[6124] logger.c: --- (10 headers 7 lines)--- Sep 3 10:06:01 DEBUG[6124] chan_sip.c: Acked pending invite 102 Sep 3 10:06:01 DEBUG[6124] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of R equest 102: Match Found Sep 3 10:06:01 VERBOSE[6124] logger.c: Found RTP audio format 0 Sep 3 10:06:01
Re: [asterisk-users] Asterisk not sending RTP
Nir Simionovich wrote: Sep 3 10:06:00 VERBOSE[6124] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), c Very simple, there is no codec being sent from the peer, thus the near end wouldn't be sending RTP. Jeremy McNamara ___ --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] UK (BT) Problem with TDM 400P
I've tried without the include, but reverted to it because it was generated like that. An ordinary phone works fine on the line and on either of the FXS sockets it gives a dialing tone (and I can see it on the asterisk console). The strange thing is that nothing whatsoever shows when I ring the line, regardless of what monitor I try. If it's any help, zttool looks like: Current Alarms: No alarms. Sync Source:Internally clocked IRQ Misses: 0 Bipolar Viol: 0 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 4/ 3/ 0 1234 TxA --- TxB --- TxC --- TxD --- RxA --- RxB --- RxC --- RxD --- (and that all stays the same when I stop or start asterisk or try dialing in). Thanks for any help Mark On 03/09/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mark Muffett wrote: It's an FXO module (I've got two FXS modules on the same card and they work - I get a dialing tone from them), and the phone line works with an ordinary phone. Have you tried not using the include? Does a normal phone attached to the line work? Does it work connected to an FXS socket (i.e. FXO - FXS)? Do you see anything when you ring the line? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE+uWrS6d5vy0jeVcRAog/AJwL6PikP+I0zaO6T4AXVRLxy8pj/ACfT0qk ly4/JGV1VyaLEsCsgdzm6cE= =TnVB -END PGP SIGNATURE- ___ --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] SIPP problem
2006/9/2, Greg Boehnlein [EMAIL PROTECTED]: On Sat, 2 Sep 2006, Diego Quintana Cruz wrote: Hi everybody, I'm trying to load-test my Asterisk PBX using SIPP, but I always getting errors, I followed the instructions given in [1] which mainly was to create the user sipp in sip.conf and the dialing plan for his context in extensions.conf I'm using Asterisk 1.0.10 Any ideas or tutorial on how using SIP? Here are my notes on the subject: http://lists.digium.com/pipermail/asterisk-dev/2006-June/021162.html I did what you have there but I'm always getting 503 Service unavailable, I don't know why. I'm also using AMPortal, do I have to configure something there? Regards, and sorry for my bad english -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --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] UK (BT) Problem with TDM 400P
Mark Muffett [EMAIL PROTECTED] wrote: I'm trying to get my TDM400P to work with a BT POT line. when I connect the FXO unit to the BT line it just makes it (the BT line) go permanently engaged. has anyone ever managed to get a TDM400P to work with a BT line and did they have any of these issues? Yes, I had a test system with a TDM400P and two FXO modules terminating two BT Featureline Compact lines. However, had is the operative word. One of the two FXO modules is now holding the line permanently off-hook - even when the system is powered down. The other module is fine, and the drivers don't see anything wrong with either module. My suspicion is that the one module is dead. I haven't had a chance to pull out the machine and play with it, though. To see if you have a similar problem, connect a phone in parallel with the FXO module. Unplug the module from the line, and check that you get a dial tone with the handset. Hang up, and plug the FXO back in. Then lift the handset again - I get a noisy line with a very quiet dial tone. If you wait long enough, the dial tone will stop and you'll hear the screamer, again very quiet. I've no idea what caused our fault. I don't know of any storms before the failure, and all the other connected equipment is fine. Nick. -- Nick Chalk . once a Radio Designer Confidence is failing to understand the problem. ___ --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 not sending RTP
Here's the funky part, it suddenly started working all of a sudden! I'm confident the carrier I'm working with simply changed something, after I bashed their heads and they claimed that everything is ok at their end. Nir S - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, September 3, 2006 4:45:08 PM GMT+0200 Subject: Re: [asterisk-users] Asterisk not sending RTP Nir Simionovich wrote: Sep 3 10:06:00 VERBOSE[6124] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), c Very simple, there is no codec being sent from the peer, thus the near end wouldn't be sending RTP. Jeremy McNamara ___ --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 -- Kind Regards, Nir Simionovich Chief Technology Officer Atelis PLC ___ --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+ser+docs
Where can I find docs on ser and asterisk intergration. Well, you could start googling around just a bit, but here you go anyway: I found these both useful: http://www.voip-info.org/wiki/view/OpenSER http://www.iptel.org/ser/doc/gettingstarted -- Victor Toofic ___ --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] UK (BT) Problem with TDM 400P
Nick Thanks. I've just tried what you suggested and get EXACTLY the same - very quiet dialtone on a noisy line, then the screamer. I suppose it must be a dodgy FXO module. Thanks again, Mark On 03/09/06, Nick Chalk [EMAIL PROTECTED] wrote: Mark Muffett [EMAIL PROTECTED] wrote: I'm trying to get my TDM400P to work with a BT POT line. when I connect the FXO unit to the BT line it just makes it (the BT line) go permanently engaged. has anyone ever managed to get a TDM400P to work with a BT line and did they have any of these issues? Yes, I had a test system with a TDM400P and two FXO modules terminating two BT Featureline Compact lines. However, had is the operative word. One of the two FXO modules is now holding the line permanently off-hook - even when the system is powered down. The other module is fine, and the drivers don't see anything wrong with either module. My suspicion is that the one module is dead. I haven't had a chance to pull out the machine and play with it, though. To see if you have a similar problem, connect a phone in parallel with the FXO module. Unplug the module from the line, and check that you get a dial tone with the handset. Hang up, and plug the FXO back in. Then lift the handset again - I get a noisy line with a very quiet dial tone. If you wait long enough, the dial tone will stop and you'll hear the screamer, again very quiet. I've no idea what caused our fault. I don't know of any storms before the failure, and all the other connected equipment is fine. Nick. -- Nick Chalk . once a Radio Designer Confidence is failing to understand the problem. ___ --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] Asterisk+ser+docs
http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration On 9/3/06, Victor Toofic [EMAIL PROTECTED] wrote: Where can I find docs on ser and asterisk intergration.Well, you could start googling around just a bit, but here you go anyway: I found these both useful: http://www.voip-info.org/wiki/view/OpenSER http://www.iptel.org/ser/doc/gettingstarted --Victor Toofic___--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] SER+Asterisk integration
Have a check through:http://www.voip-info.org/wiki-NAT+and+VOIPhttp://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
RegardsRobOn 03/09/06, Siqhamo Sifo [EMAIL PROTECTED] wrote:
I have ser sitting on my iptables nat boxand my asterisk box on the lan. Ser does forwarding so that any requests (register,invite,ack,...) tothe nat box at 5060 r sent to my asterisk box on thelan .I can register
from outsideto my asterisk box but there is only one way audio , reason being thatwhen the asterisk box sends a sip packet whith session description the sdppart of the sip packet is not natted .I have tried the following:
if(src_ip==10.0.0.0/255.0.0.0){force_rtp_proxy(); encode_contact(enc_prefix,wanip); sdp_mangle_ip(
10.0.0.0/255.0.0.0,wanip);};and it does not work because my ethernet dump shows that the contact insdp is
not mangled.___--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] SIPP problem
On Sun, Sep 03, 2006 at 10:03:32AM -0500, Diego Quintana Cruz wrote: 2006/9/2, Greg Boehnlein [EMAIL PROTECTED]: On Sat, 2 Sep 2006, Diego Quintana Cruz wrote: Hi everybody, I'm trying to load-test my Asterisk PBX using SIPP, but I always getting errors, I followed the instructions given in [1] which mainly was to create the user sipp in sip.conf and the dialing plan for his context in extensions.conf I'm using Asterisk 1.0.10 Any ideas or tutorial on how using SIP? Here are my notes on the subject: http://lists.digium.com/pipermail/asterisk-dev/2006-June/021162.html I did what you have there but I'm always getting 503 Service unavailable, I don't know why. I'm also using AMPortal, do I have to configure something there? Do you use sipp as a standaalone service, or do you also need an Asterisk to originate calls? If the former, An Asterisk installation is not really required and shouldn't matter, anyway. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] AgentCallBackLogin and cdrupdate
Hi All The option cdrupdate=yes is supposed to update the CDR records to reflect the agent that is logged in and not the static extension that is making / receiving calls. This is not working at the moment? Incoming calls get updated in the CDR correctly, however outgoing calls from the agent still show as the static extension. Kind Regards Garth -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO MSN:[EMAIL PROTECTED] Web:www.bitco.co.za ___ --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] Queue timeout problems
Hi Guido - Evidently I needed to add a timeout to the queue itself. Thanks, Brian On 9/3/06, Guido Hecken [EMAIL PROTECTED] wrote: -Ursprüngliche Nachricht- Von: Mr. Jones [mailto:[EMAIL PROTECTED] Gesendet: Sonntag, 3. September 2006 06:10 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Queue timeout problems Thanks Guido - I tried that and still have the same problem. The call never seems to leave the queue. Any other ideas? Hmm, to have a closer look on the problem, one could do the following Activate debugging, error and verbose logging in logger.conf by having a line like this: console = notice,warning,error,debug,verbose Open the cli and do a logger reload set verbose to 5 or even 255 Initiate a call to the queue and watch for errors/informations. Perhaps, define a context named test and put a really simple command in it. Something like this [test] exten = 120,1,Answer() exten = 120,2,Playback(some-sound-file) exten = 120,3,Hangup Change your queue to call this context in the second priority. Also have a closer look on your include commands in the dialplan... Normally an extensions reload on the cli should activate the changes to the dialplan, but with a restart now you should be save. Good luck Guido ___ --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] SER+Asterisk integration
have a look at the nathelper examples in SER distribution. This is from
an rather old installation of mine.
--
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received
if (nat_uac_test(3)) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method == REGISTER || ! search(^Record-Route:)) {
xlog(L_ERR, LOG: Someone trying to register from private
IP, rewriting\n);
# This will work only for user agents that support symmetric
# communication. We tested quite many of them and majority i
s
# smart enough to be symmetric. In some phones it takes a co
nfiguration
# option. With Cisco 7960, it is called NAT_Enable=Yes, with
kphone it is
# called symmetric media and symmetric signalling.
fix_nated_contact(); # Rewrite contact with source IP of sig
nalling
if (method == INVITE) {
fix_nated_sdp(1); # Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6);# Mark as NATed
};
};
..
# if client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
--
___
--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] SIPP problem
For the record, we use a very similar test environment for Asterisk on the Blackfin: * Astersik 1.0.11 (latest Rapid stable debs) - or 1.2.9/1.2.10 from our unstable debs * Diego does most of the job ;-) Anyway, I suggest that you re-read that page. You basically need to alightly eit the supplied sip.conf to match your settings, and also play a bit with sipp (package sip-tester on Debian). Yes, it was my mistake, i create the extension with the context from-internal and everything went fine, now I'm having another problem, which is that I'm calling the echo-test extension, but asterisk hangs me 30 seconds later because sipp is not sending any RTP data. Any ideas on how to fix this. The demo context which is mentioned in [1] doesn't work. [1] http://www.rowetel.com/ucasterisk/ucasterisk.html Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --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] What I always get asked in SME * deployments
Some phones have the BLF feature. You can see on the phone who is and who is not on the phone. With the polycom's you need to get a side car. With the snom's you can use the buttons on the phone itself. When ever we do a roll out of Asterisk in a small business environment replacing an old key system or legacy PBX the receptionist always asks us, "How do I know if someone is on a call before transferring them?". My typical answer is "why do you need to know, just do an attended transfer and if they can take the call they will, if they can't just tell the caller the person is busy". If the receptionist insists on "knowing" we give them FOP.Has anyone out there devised a better way to let a receptionist "know if someone is on a call"? ___ --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] Polycoms, Attended Transfer and Canreinvite = yes
Well personally I am just glad I wasn't the only one seeing the problem. As much as I don't like the place 100% of the blame on something unless I fully know what is going on, in this case Asterisk, but I couldn't see any solution but a bug. Personally I wouldn't mind testing out the branch, but I know my boss, isn't so trusting. How stable are the SVN branches, at least in terms of justification for taking the system down to install it? Or is there an easier way to test? Thanks, Kevin Kevin P. Fleming wrote: - Richard Scobie [EMAIL PROTECTED] wrote: Dave Fullerton wrote: I just verified it here as well. Running Asterisk 1.2.11 and two polycom I'll throw in a "me too" here, with the addition that it also occurs with "canreinvite=no". There were multiple problems in this area, introduced since Asterisk 1.2.9 was released. We believe that with today's commits in SVN branch-1.2 they are cured, so it would help us greatly if could download SVN branch-1.2 and try it out on your system to see if it solves your issue. I apologize for how this crept into the code base... it should not have happened, and we are taking steps to ensure that future changes in the release branch don't cause regressions like this. ___ --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] Polycoms, Attended Transfer and Canreinvite = yes
The svn branch-1.2 is very stable, probably more stable then the rpms and other distro's out there, as fixes are applied when problems are identified and corrected. Sometime later, the svn branch-1.2 is used to create packages. Kevin Smith wrote: Well personally I am just glad I wasn't the only one seeing the problem. As much as I don't like the place 100% of the blame on something unless I fully know what is going on, in this case Asterisk, but I couldn't see any solution but a bug. Personally I wouldn't mind testing out the branch, but I know my boss, isn't so trusting. How stable are the SVN branches, at least in terms of justification for taking the system down to install it? Or is there an easier way to test? Thanks, Kevin Kevin P. Fleming wrote: - Richard Scobie [EMAIL PROTECTED] wrote: Dave Fullerton wrote: I just verified it here as well. Running Asterisk 1.2.11 and two polycom I'll throw in a me too here, with the addition that it also occurs with canreinvite=no. There were multiple problems in this area, introduced since Asterisk 1.2.9 was released. We believe that with today's commits in SVN branch-1.2 they are cured, so it would help us greatly if could download SVN branch-1.2 and try it out on your system to see if it solves your issue. I apologize for how this crept into the code base... it should not have happened, and we are taking steps to ensure that future changes in the release branch don't cause regressions like this. ___ --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] PBX - VoIP migration
Morning (o; What would give me less headache for integrating a Nortel PBX to VoIP? a) Hook up with a Cisco which handles the SIP stuff and E1 to telco failover? b) Hook it up to an asterisk box instead? If I would go with plan (b)...is there an option I can sort of pipe through the E1 trunk coming from local telco within asterisk to the PBX but also start hooking up SIP accounts to it? cheers rick ___ --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] Polycoms, Attended Transfer and Canreinvite = yes
Kevin P. Fleming wrote: if could download SVN branch-1.2 and try it out on your system to see if it solves your issue. Is there a Wiki page or similar describing how to checkout SVN for Asterisk? Also, will I need to checkout and compile SVN versions of Zaptel/Libpri/Addons (as I use all three)? Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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] Roundrobin not working on PRI
Can somebody send me a sample from their extension.conf to do the above mentioned thing, i.e. handling DIDs on PRI. This is the first time I am dealing with PRI, previously I always used SIP DIDs and had no problem at all. ___ --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] Blind transfer 3/4 digits
Hi all, Ronald, if you are using #, try adjusting the featuredigittimeout parameter in features.conf.This is the max time between digits for feature activation. If is small, * could dial the wrong number, in your case 601 instead of 6014. I think that you are not using # while your are using snom, because you said that you needed to dial # in order to finish the transfer (this it's no necessary for *). Or snom is catching the # and driving the transfer. fabay -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Ronald Wiplinger Enviado el: Sábado, 02 de Septiembre de 2006 10:40 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Blind transfer 3/4 digits Tim St. Pierre wrote: Are you using # to transfer? If so, it's not sending it as a new call, it's just sending asterisk digits using whatever DTMF mode. Asterisk parses these based on a first match in the dialplan. Make sure that the longer extension numbers are loaded first in the dialplan. That is a good thought. I can remember that the docs said that you cannot force the order of the dialplan, except with includes. I will try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? bye Ronald -Tim On September 2, 2006 20:12, Ronald Wiplinger wrote: Kevin Smith wrote: Dialing a number and transferring a number are two different things. And no offense, you are not really providing a lot of details along with your problem. So you can dial the numbers but not transfer from one to the other. I was not thinking that it would be too much difference. Therefore I also do not know what more info could help to distinguish the problem. I hardly can post my entire configuration. What does the CLI say when you try the transfer? That would provide a lot of information that could clue you in to what is going on. You hit another problem with that. I hardly see here anything anymore. The messages fly by so fast, Especially annoying messages: chan_sip.c:10888 handle_request_register: Registration from 'sip:192.168.250.20' failed for '192.168.250.244' - Username/auth name mismatch -- Got SIP response 486 Busy Here back from 192.168.250.244 -- Got SIP response 400 Bad Request back from xx.xx.xx.126 NOTICE[5936]: chan_sip.c:9600 handle_response_register: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3) . It would be nice to filter the CLI for such investigation for a moment. What type of phones are you using? Some phones have the ability to pattern match and wait for a certain number of seconds before sending the number to asterisk. For example. On our Polycom phones a user has 3 seconds (between digits) to enter in 10 digits. This could be where most of your problem is. That is a very good point and I will contact the manufacturer of these no-name phones. My guess the problem lies with the Phones, not Asterisk form the information you provided. I disagree with that! Why Asterisk treats dialing and transfer different. That makes not really sense, does it? bye Ronald Kevin Ronald Wiplinger wrote: David Gagnon wrote: Ronald, You seem to be a little bit angry about VoIP. If so, I could give you my old Nortel system. Does this would make you happy? David David, I am not angry about VoIP, but please send my your old Nortel system ! I just do not understand why I can DIAL 601 and 6014, but not use blind transfer. Is the question too difficult? I am sure there is somewhere a switch to say, wait two seconds (as for dialing) before you assume it is a complete number. It is also strange that snom phone can do it correct, because it uses the ok key. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 2 septembre 2006 04:20 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Anthony Rodgers wrote: With respect, the problem is with your numbering plan.. This answer is therefore totally nonsense !!! (With all respect!!!) Both answers have actually not lead to any step further, but to more messages. I use to refer to such answers as NON-ANSWERS. Please only reply if and really only if you know a solution for the problem! Thanks for your understanding. bye Ronald - again, I am not angry at all. WHERE do you see a problem in the numbering plan? I see the problem in ASTERISK, because it does not wait for the last digit!!! Where can I set that it waits for it? The beauty on voip IS that you can have different length and overlapping, bye Ronald CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get
Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Avi Miller wrote: Is there a Wiki page or similar describing how to checkout SVN for Asterisk? Also, will I need to checkout and compile SVN versions of Zaptel/Libpri/Addons (as I use all three)? Replying to myself to say that I've found Digium's instructions and I'm testing SVN on my test server now. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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] Polycoms, Attended Transfer and Canreinvite = yes
Avi Miller wrote: Replying to myself to say that I've found Digium's instructions and I'm testing SVN on my test server now. :) And again to say that it seems work just fine with the SVN code. Thanks Kevin! -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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] PBX - VoIP migration
I have done quite a few E1 - Asterisk - PABX setups, and they are quite flexible. Check your figures though. I set up at a site, and after the event the manager realised that it cost the same as moving the office over to Asterisk completely. (due to the expense of having work done on their NEC system as well...) regards, PaulH On Mon, 2006-09-04 at 00:12 +0200, Richard Klingler wrote: Morning (o; What would give me less headache for integrating a Nortel PBX to VoIP? a) Hook up with a Cisco which handles the SIP stuff and E1 to telco failover? b) Hook it up to an asterisk box instead? If I would go with plan (b)...is there an option I can sort of pipe through the E1 trunk coming from local telco within asterisk to the PBX but also start hooking up SIP accounts to it? cheers rick ___ --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] Polycoms, Attended Transfer and Canreinvite = yes
On Mon, 2006-09-04 at 08:44 +1000, Avi Miller wrote: Avi Miller wrote: Is there a Wiki page or similar describing how to checkout SVN for Asterisk? Also, will I need to checkout and compile SVN versions of Zaptel/Libpri/Addons (as I use all three)? Replying to myself to say that I've found Digium's instructions and I'm testing SVN on my test server now. :) SVN is goodand AJAM looks great :) later, PaulH ___ --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] Please help route incoming PSTN calls to Asterisk
I have a working Asterisk 1.2.5 system with SPA-3000 setup with the SPA3000 Configuration Wizard for Asterisk from Voxilla.com. I can make outbound calls from the Sipura POTS phone (not sure they are actually going through the Asterisk box) but cannot get inbound calls from the outside. Problem is I get no apparent response from Asterisk when PSTN calls come in, although POTS phones on the PSTN line ring ok. I'm pretty sure something is wrong in my configuration but I can't see what is wrong after a lot of web and book searching. The system does me little good if I can't at lease _receive_ calls over the POTS line which is where most of my calls come from. The rest is the part of sip.config and extensions.conf. I hope someone will give me tips to get it ringing. sip.conf: --- [telasip-gw] ; Gateway ; context=telasip-in type=friend qualify=200 host=gw3.telasip.com username=lalkoff secret=xx insecure=very canreinvite=no callerid=Larry Alkoff 5123011411 nat=yes [200] ; Sipura Line 1 outbound to PSTN type=friend host=dynamic context=home secret=xxx mailbox=200 dtmfmode=rfc2833 disallow=all allow=ulaw [201] ; Sipura forward PSTN inbound to Asterisk ; If you're using Asterisk, this goes into the Incoming ; settings for your Trunk type=friend host=dynamic ; If using [EMAIL PROTECTED], change the below line to context=from-internal context=home secret=7883982 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very [pstn-spa3k] ; Asterisk VOIP outbound to PSTN ; If you're using Asterisk, this section goes into the ; Outgoing Settings for your trunk. type=peer auth=md5 host=192.168.0.41 port=5061 secret=xxx username=asterisk fromuser=asterisk dtmfmode=rfc2833 context=home insecure=very --- end of sip.conf extensions.conf: -- [home] ; Sipura forward PSTN inbound to Asterisk exten = 200,1,Ringing exten = 200,2,Dial(SIP/200,20,T) exten = 200,3,Voicemail(u200) exten = 200,4,Hangup exten = 911,1,Dial(SIP/[EMAIL PROTECTED],60,) exten = 911,2,Congestion exten = _XXX,1,SetCallerID(512-301-1410) exten = _XXX,2,Dial(SIP/[EMAIL PROTECTED],60,) exten = _XXX,3,Congestion ;;exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED],60,) ;;exten = _XXX,2,Congestion exten = _1800XXX,1,Dial(SIP/[EMAIL PROTECTED],60,) exten = _1800XXX,2,Congestion exten = _1888XXX,1,Dial(SIP/[EMAIL PROTECTED],60,) exten = _1888XXX,2,Congestion exten = _1877XXX,1,Dial(SIP/[EMAIL PROTECTED],60,) exten = _1877XXX,2,Congestion exten = _1866XXX,1,Dial(SIP/[EMAIL PROTECTED],60,) exten = _1866XXX,2,Congestion --- end of extensions.conf - I'd most gratefully appreciate any help you could give me. Larryalk -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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] Help with blind transfer
This is not the list for [EMAIL PROTECTED]. For questions about [EMAIL PROTECTED] functionality, they have their own mailing list. bp On 9/3/06, George A. Roberts IV [EMAIL PROTECTED] wrote: No one has any ideas? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of George A. Roberts IV Sent: Friday, September 01, 2006 10:37 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Help with blind transfer Hello all, Using [EMAIL PROTECTED] and are using its Follow Me functionality to bounce calls to my cell phone when I'm not in the office. Would like to be able to use the blind transfer functionality from my cell phone when I receive a call in from Asterisk but am not having much luck getting it to work… I can press ## (that's what it's set to in features.conf) and get the "Transfer" prompt from Alison and the dialtone. But no matter what I punch in, it seems that Asterisk is only getting the first digit I press. -- Unable to find extension '*' in context 'from-internal' -- Playing 'pbx-invalid' (language 'en') -- Stopped music on hold on SIP/801-b190 -- Started music on hold, class 'default', on SIP/801-b190 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '8' in context 'from-internal' -- Playing 'pbx-invalid' (language 'en') The first one there I tried to punch in *801 to transfer to voicemail. The second one I punched in 802 to transfer to another extension. Each time, it's only getting the first digit. Anyone seen this before? Any thoughts? Thanks! Regards, George A. Roberts IV President CEO, Interjuncture Corp. http://www.interjuncture.com/ ___--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] Help with blind transfer
This isnt [EMAIL PROTECTED] functionality. Its basic Asterisk functionality. George From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Piper Sent: Sunday, September 03, 2006 9:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with blind transfer This is not the list for [EMAIL PROTECTED]. For questions about [EMAIL PROTECTED] functionality, they have their own mailing list. bp On 9/3/06, George A. Roberts IV [EMAIL PROTECTED] wrote: No one has any ideas? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of George A. Roberts IV Sent: Friday, September 01, 2006 10:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help with blind transfer Hello all, Using [EMAIL PROTECTED] and are using its Follow Me functionality to bounce calls to my cell phone when I'm not in the office. Would like to be able to use the blind transfer functionality from my cell phone when I receive a call in from Asterisk but am not having much luck getting it to work I can press ## (that's what it's set to in features.conf) and get the Transfer prompt from Alison and the dialtone. But no matter what I punch in, it seems that Asterisk is only getting the first digit I press. -- Unable to find extension '*' in context 'from-internal' -- Playing 'pbx-invalid' (language 'en') -- Stopped music on hold on SIP/801-b190 -- Started music on hold, class 'default', on SIP/801-b190 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '8' in context 'from-internal' -- Playing 'pbx-invalid' (language 'en') The first one there I tried to punch in *801 to transfer to voicemail. The second one I punched in 802 to transfer to another extension. Each time, it's only getting the first digit. Anyone seen this before? Any thoughts? Thanks! Regards, George A. Roberts IV President CEO, Interjuncture Corp. http://www.interjuncture.com/ ___ --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 smime.p7s Description: S/MIME cryptographic signature ___ --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] Help with blind transfer
Some additional information: It looks like its not even waiting for any additional DTMF signals as it immediately tries to find the extension as soon as it gets the first digit: Sep 3 22:54:42 VERBOSE[15732] logger.c: -- Playing 'pbx-transfer' (language 'en') Sep 3 22:54:44 DEBUG[15738] chan_sip.c: * Detected inband DTMF '*' Sep 3 22:54:48 VERBOSE[15732] logger.c: -- Unable to find extension '*' in context 'from-internal' Sep 3 22:54:48 VERBOSE[15732] logger.c: -- Playing 'pbx-invalid' (language 'en') Sep 3 22:54:52 VERBOSE[15732] logger.c: -- Stopped music on hold on SIP/801-b190 I was watching this log as it was generated I dialed *801 but it only got the first DTMF. This was tested by picking up my SIP handset, dialing , then dialing 801 which did an outbound call to my cell phone. When I answered my cell phone, I pressed # and was issued the prompt Transfer I dialed *801, but as I said above its not getting the additional DTMF tones. If I dial *in* to the system from my cell phone and dial 801 it will send me to the SIP extension. From there, I can successfully hit # and transfer the call to *801 with no problems. It only seems to be when the call is transferred to an external phone. This isnt an issue with the transfer system not finding the extension due to improper context or something this is an issue with Asterisk recognizing DTMF tones during transfer. Any thoughts? George From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George A. Roberts IV Sent: Sunday, September 03, 2006 10:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Help with blind transfer This isnt [EMAIL PROTECTED] functionality. Its basic Asterisk functionality. George From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Piper Sent: Sunday, September 03, 2006 9:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with blind transfer This is not the list for [EMAIL PROTECTED]. For questions about [EMAIL PROTECTED] functionality, they have their own mailing list. bp On 9/3/06, George A. Roberts IV [EMAIL PROTECTED] wrote: No one has any ideas? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of George A. Roberts IV Sent: Friday, September 01, 2006 10:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help with blind transfer Hello all, Using [EMAIL PROTECTED] and are using its Follow Me functionality to bounce calls to my cell phone when I'm not in the office. Would like to be able to use the blind transfer functionality from my cell phone when I receive a call in from Asterisk but am not having much luck getting it to work I can press ## (that's what it's set to in features.conf) and get the Transfer prompt from Alison and the dialtone. But no matter what I punch in, it seems that Asterisk is only getting the first digit I press. -- Unable to find extension '*' in context 'from-internal' -- Playing 'pbx-invalid' (language 'en') -- Stopped music on hold on SIP/801-b190 -- Started music on hold, class 'default', on SIP/801-b190 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '8' in context 'from-internal' -- Playing 'pbx-invalid' (language 'en') The first one there I tried to punch in *801 to transfer to voicemail. The second one I punched in 802 to transfer to another extension. Each time, it's only getting the first digit. Anyone seen this before? Any thoughts? Thanks! Regards, George A. Roberts IV President CEO, Interjuncture Corp. http://www.interjuncture.com/ ___ --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 smime.p7s Description: S/MIME cryptographic signature ___ --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 calling through FWD?
Hi all, I have been researching a dialing problem I am having with FWD. I followed their IAX2 config notes, and I can receive calls from my brother from FWD, and all the echo tests, call me services work. But I cannot call him. -- Executing SetCallerID(SIP/4003-9de6, Nick Ellson) in new stack -- Executing Dial(SIP/4003-9de6, IAX2/776754:scrubbed@iax2.fwdnet.net/snipped|60|r) in new stack -- Called 776754:scrubbed@iax2.fwdnet.net/snipped -- Call accepted by 192.246.69.186 (format ulaw) -- Format for call is ulaw -- IAX2/192.246.69.186:4569-2 is busy -- Hungup 'IAX2/192.246.69.186:4569-2' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Congestion(SIP/4003-9de6, ) in new stack == Spawn extension (default, 393number snipped, 3) exited non-zero on 'SIP/4003-9de6' This is pretty much just what a few others from the FWD forums have posted with no real response. Has any one of you also had this problem with FWD? Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --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
