Re: [Asterisk-Users] res_motv: Request for Comment
It might be nice if safe_asterisk (or some part of asterisk) e-mailed a backtrace of the last asterisk .core file to Digium so they can see what causes Asterisk to core dump. I've not had asterisk crash in that way, but it might be nice for Digium. On Tue, 6 Apr 2004, Mark Spencer wrote: I've been considering the nature of Asterisk, its security, the bug tracker, and more... And i've come up with an interesting idea: A message of the version. The idea is that Asterisk has a compile time 32-bit unsigned int version which is incremented whenever some major new bug is fixed. When Asterisk starts up (and periodically, maybe once per day), it sends a packet with the version number to a server at Digium, along with a message level (INFO,MINOR,MAJOR,CRITICAL) and the Digium server replies (if it receives the packet, if not, it might get sent again in a day) with any INFO, MINOR, MAJOR, or CRITICAL messages which are associated with that version of the code. In this way, an asterisk administrator could easily see if there were any major issues, critical security updates, etc, that his system might need to be updated for. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Passing DTMF
Eric Wieling wrote: On Tue, 2004-04-06 at 12:29, Brian Rathman wrote: Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco AS5300 with * in the media stream. Unfortunately, the only way I can get the calls to connect is with t or T at the end of the Dial() statement and then that picks off the dtmf digits. I have tried the canreinvite=yes on both the phone peer and the gateway peer and I still have to add the T to the Dial statement to make the call complete. Any suggestions??? cantrinvite=yes tells asterisk to, if it can, remove itself from the media stream. T and t and r and many other Dial options tells Asterisk to stay in the media stream so it can listen to the DTMF. None of this has ANYTHING to do with passing DTMF between the two endpoints (except of course passing # for t or T). If you cannot pass DTMF between the two endpoints then something ELSE is wrong. Maybe you are trying to use inband DTMF with a compressed codec. Inband DTMF will only work with ulaw or alaw codecs. ...or the problem is, as hinted, that Asterisk sends a short dtmf. Regardless of what it receives into the sip channel, Asterisk sends a 250 ms DTMF signal out (if my memory is correct). You can check in chan_sip.c The dtmf setting sets what Asterisk sends to that peer/user. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Passing DTMF
...or the problem is, as hinted, that Asterisk sends a short dtmf. This is a bug in 7.2 release. It's setting the duration of RFC2833 digits to as low as 30 msec which is definitely not enough. To fix, change the following in rtp.c, function ast_rtp_senddigit() /* Make duration 240 */ rtpheader[3] |= htonl((240)); The duration is based on 8000 Hz clock so 240 equals to 30 msec. Use at least 1200 for 150 msec tones or even more as needed. With kind regards, M. -- Marian Durkovic network manager Slovak Technical University Tel: +421 2 524 51 301 Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351 812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] res_motv: Request for Comment
Hi, -Original Message- Now, of course, any time you put a call home feature in, there are people who will be concerned about privacy. Clearly it will be able to be disabled, but I want to run my idea about deployment by everyone here and see if you guys had some ideas. The idea would be that *new* installs (make samples) would have the feature turned on for MAJOR level by default, and that any existing install (e.g. /etc/asterisk/sip.conf exists, but not /etc/asterisk/motv.conf) would have the file created at the next make install based upon prompting the installer. Sounds like a nice feature. Things to consider: - What if someone has their own development tree to work with - the 'call home server' should be configurable ? - As a security consideration, sending the local version might not be wise - what if the call home server is being dns-spoofed ? An intruder might get relevant version info... Downloading a changes matrix might be wise (maybe related to 'last time checked' rather than version) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Passing DTMF
Marian Durkovic wrote: ...or the problem is, as hinted, that Asterisk sends a short dtmf. This is a bug in 7.2 release. It's setting the duration of RFC2833 digits to as low as 30 msec which is definitely not enough. To fix, change the following in rtp.c, function ast_rtp_senddigit() /* Make duration 240 */ rtpheader[3] |= htonl((240)); The duration is based on 8000 Hz clock so 240 equals to 30 msec. Use at least 1200 for 150 msec tones or even more as needed. Please add this to bugs.digium.com so it can be changed in the CVS tree. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Callerid + Zaphfc
Hi all, i have an ISDN Phone connected to an HFC-S based card, all works fine but is i call the Phone from a SIP User Agent or over PSTN Line the Phones Display shows the correct CallerID but with a leading 0 . I cant find this in the config files, how can is solve this? Dialing Out with the ISDN Phone transmitts the correct callerid. Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_motv: Request for Comment
Hi a) The idea itself -- is it a good one or is it stupid? great idea. could be very useful if you don't have much time to track/test cvs version and/or the bugtracker b) The way to make it deployed without sneaking a call home in on anybody that doesn't want it? make it off by default, providing infos on how to enable it. In this way you don't have to worry about user complaints about privacy (hey, you've turned on! isn't that by default), Also not all systems could have a open internet connection... so sending infos is impossible at all. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_motv: Request for Comment
I'd like to give this one 10 thumbs down. IMHO a bad idea, a nasty little bad idea.. evil, spawn of Satan. If this were implemented the first job of a new update would be to rip it out and flush it down the nearest toilet. I can only wait until we see M$ like activation implemented... oh the joy... It would be much better just to have the information present on either the Digium site or some other location. I see little point in wasting your valuable time doing something like this when there are so many outstanding issues and feature requests that could offer more. Andy *** REPLY SEPARATOR *** On 06/04/2004 at 22:31 Mark Spencer wrote: I've been considering the nature of Asterisk, its security, the bug tracker, and more... And i've come up with an interesting idea: A message of the version. The idea is that Asterisk has a compile time 32-bit unsigned int version which is incremented whenever some major new bug is fixed. When Asterisk starts up (and periodically, maybe once per day), it sends a packet with the version number to a server at Digium, along with a message level (INFO,MINOR,MAJOR,CRITICAL) and the Digium server replies (if it receives the packet, if not, it might get sent again in a day) with any INFO, MINOR, MAJOR, or CRITICAL messages which are associated with that version of the code. In this way, an asterisk administrator could easily see if there were any major issues, critical security updates, etc, that his system might need to be updated for. Now, of course, any time you put a call home feature in, there are people who will be concerned about privacy. Clearly it will be able to be disabled, but I want to run my idea about deployment by everyone here and see if you guys had some ideas. The idea would be that *new* installs (make samples) would have the feature turned on for MAJOR level by default, and that any existing install (e.g. /etc/asterisk/sip.conf exists, but not /etc/asterisk/motv.conf) would have the file created at the next make install based upon prompting the installer. Any feedback on: a) The idea itself -- is it a good one or is it stupid? b) The way to make it deployed without sneaking a call home in on anybody that doesn't want it? Thanks! Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone registering problem
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote: download ethereal and take a peek at the packets on the wire. Without something like that, no one is really going to be able to help you. Do you mean then that my SIP trace displayed at kphone looks otherwise OK -- that the REGISTER request that kphone's sending out looks alright? Is there a single good resource describing the SIP protocol specification so I know what I should be looking for? thanks, Richard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone registering problem
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote: download ethereal and take a peek at the packets on the wire. Without something like that, no one is really going to be able to help you. Do you mean then that my SIP trace displayed at kphone looks otherwise OK -- that the REGISTER request that kphone's sending out looks alright? Is there a single good resource describing the SIP protocol specification so I know what I should be looking for? One good reference is: http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programming_reference_guide_book0 9186a0080080221.html Since you indicated asterisk was not showing any activity from that register, it is highly likely the register activity isn't going to where you think it should, or something like that. I'd suspect that you really don't need the sip protocol reference noted above, just a clue from the packet trace what is really happening on the wire. SIP packets (on the wire) contain a substantial amount of plain text, so looking at two or three packets to see what is really happening doesn't require very much technical understanding or references. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Friends and MySql
Alex Lopez wrote: What is the difference b/w USE_MYSQL_FRIENDS=1 and USE_SIP_MYSQL_FRIENDS=1 Not sure ;) Am I to think that this replaces the entrys in sip.conf for the registering clients?? Yes If so, I am hosed as I cannot get a ATA-186 to register via MySql, but if I leave the config in sip.conf all is well. Could someone send me one record from their sipfriends table that works??? http://voip-info.org/wiki-Asterisk+sip+mysql+peers I see that there is no place to specify nat=yes, host=dynamic, etc. in the table, or am I just barking up the wrong tree. http://bugs.digium.com/bug_view_page.php?bug_id=0001086 F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone registering problem
On Wednesday 07 April 2004 09:24, Richard Airlie wrote: On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote: download ethereal and take a peek at the packets on the wire. Without something like that, no one is really going to be able to help you. I thought I'd chime in here with a packet dump from ethereal, since I'm also using KPhone 4.02 and I feel there's maybe a bug in its registration code, since I remember using KPhone 3.xx and having no problems. However, I would appreciate it if someone more experienced could cast their eyes over this to make sure I'm not doing something stupid :) My Asterisk sip.conf is simply: [general] port = 5060 bindaddr = 0.0.0.0 context = default [myphone] type=friend host=dynamic context=default qualify=1000 And the KPhone configuration is set as follows: User part of SIP URL: myphone Host part of SIP URL: gdh Outbound Proxy: tel 'gdh' is the name of my local PC and resolves to 194.200.209.137 'tel' is the remote Asterisk server at 213.2.4.46. I have made sure there is no firewall issue between the two by two catch-all iptables -A FORWARD -s 194.200.209.137 -d 213.2.4.46 -j ACCEPT and iptables -A FORWARD -d 194.200.209.137 -s 213.2.4.46 -j ACCEPT at the firewall on each side. I've attached a gzipped snippet from ethereal in libpcap format of the failing registration. If the list doesn't permit attachments, it's also available at http://bum.net/sip.cap Cheers, Gavin. sip.cap.gz Description: GNU Zip compressed data
Re: [Asterisk-Users] res_motv: Request for Comment
Interesting idea, but needs some refinement... Very few Asterisk installations are alike. I have a couple of FreeBSD asterisk installations without any zaptel stuff, without ISDN, without MGCP, Skinny and a lot of other modules stripped out. If any of those modules have a MAJOR bug, it's not my problem. However on the Linux-based Asterisk I have in other installations, a MAJOR bug in zap or libpri would be a cause of concern. So we need to ship a reply from the service informing the installation of what modules and which versions of those modules are having a problem. And in some cases, combinations of modules that may cause problems... It needs an architecture and coders of both the client and the server. After that, a system to make decisions based on bug reports that sets the flags. Even though this is not a simple task, I agree that it would be useful as more and more Asterisks is installed in mission-critical environments. Just my 10 cents... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN calls do NOT hang up
Hi all,
In myAsterisk setup, incoming calls through
Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be
terminated properly after hangup. However, when callswere forwarded to
voicemail, after recording hangup the PSTN callsand ciscoFXO
port remained connectedunless cisco port was manually shut/no shut. # key
used to hang up the call did NOT help. Did anyone experience the same
problem??
--
sip*CLI
-- Executing
Answer("SIP/-0811b4b8", "") in new stack -- Executing
Wait("SIP/-0811b4b8", "1") in new stack -- Executing
VoiceMail("SIP/-0811b4b8", "u6917") in new stack --
Playing 'voicemail/default/6917/unavail' (language 'en')
-- Playing 'vm-intro' (language 'en') -- Playing 'beep'
(language 'en') -- x=0, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: gsm,
0x81254f8 -- x=1, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav49,
0x80fb178 -- x=2, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav,
0x811af70 -- Playing 'vm-msgsaved' (language
'en') -- Executing Hangup("SIP/-0811b4b8", "") in new
stack == Spawn extension (sip, 6917, 4) exited non-zero on
'SIP/-0811b4b8'sip*CLI
---
cisco#sh voice call
1/0/1 vtsp level
0 state = S_CONNECTvpm level 1 state = FXOLS_CONNECT vpm level 0 state =
S_UP
--
dial-peer voice 999
voipdestination-pattern 8...session protocol
sipv2session target ipv4:10.1.1.1:5065session transport
udpcodec g711ulawno vad!
exten =
6917,1,Answerexten = 6917,2,Wait(1)exten =
6917,3,VoiceMail(u${EXTEN})exten = 6917,4,Hangup
Thanks.
Ben
Re: [Asterisk-Users] PSTN calls do NOT hang up
Hi,
Asterisk either need to know when the remote caller ends his call,
or it must detect the silence.
Simplest solution is to activate silence detection, see
voicemail.conf.
You may need to do some testing to get the proper
silencethreshold setting.
Also search the archive, this is a often discussed issue...
http://mharc.lists.openservices.ca/archives/html/asterisk-users/
/Stig
At 17:25 2004-04-07 +0800, you wrote:
Hi all,
In my Asterisk setup, incoming calls
through Cisco PSTN gateway to Asterisk extensions sounds work fine. All
calls can be terminated properly after hangup. However, when calls were
forwarded to voicemail, after recording hangup the PSTN calls and
cisco FXO port remained connected unless cisco port was manually shut/no
shut. # key used to hang up the call did NOT help. Did anyone experience
the same problem??
--
sip*CLI
-- Executing
Answer(SIP/-0811b4b8, ) in new stack
-- Executing Wait(SIP/-0811b4b8,
1) in new stack
-- Executing VoiceMail(SIP/-0811b4b8,
u6917) in new stack
-- Playing 'voicemail/default/6917/unavail' (language
'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: gsm,
0x81254f8
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav49,
0x80fb178
-- x=2, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav,
0x811af70
-- Playing 'vm-msgsaved' (language 'en')
-- Executing Hangup(SIP/-0811b4b8,
) in new stack
== Spawn extension (sip, 6917, 4) exited non-zero on
'SIP/-0811b4b8'
sip*CLI
---
cisco#sh voice call
1/0/1
vtsp level 0 state = S_CONNECTvpm level 1
state = FXOLS_CONNECT vpm level 0 state = S_UP
--
dial-peer voice 999 voip
destination-pattern 8...
session protocol sipv2
session target ipv4:10.1.1.1:5065
session transport udp
codec g711ulaw
no vad
!
exten = 6917,1,Answer
exten = 6917,2,Wait(1)
exten = 6917,3,VoiceMail(u${EXTEN})
exten = 6917,4,Hangup
Thanks.
Ben
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[Asterisk-Users] indications.conf for Portugal
Title: indications.conf for Portugal Does someone have the settings for 'indications.conf' in Portugal? Thank you, Pedro Goncalves -- Pedro Goncalves PT Inovação SA - Pólo do Porto Largo de Mompilher, 22 - 4º 4050-392 Porto - Portugal Phone: +351 222079329 Email: [EMAIL PROTECTED] --
Re: [Asterisk-Users] SIP phone registering problem
First pass through the trace indicates all udp packets originating from 194.200.209.137 have incorrect checksums. However, the asterisk machine acknowledged the initial register packet with a 100 Trying, therefore it must be ignoring udp checksums. (Still curious why incorrect checksums are generated consistently.) It would appear that asterisk accepted the Register (with the 200 OK), but then .137 attempts another Register four seconds later. Might try type=user (instead of type=friend) and host=ipaddr to bypass the register function and see if that works. On Wednesday 07 April 2004 09:24, Richard Airlie wrote: On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote: download ethereal and take a peek at the packets on the wire. Without something like that, no one is really going to be able to help you. I thought I'd chime in here with a packet dump from ethereal, since I'm also using KPhone 4.02 and I feel there's maybe a bug in its registration code, since I remember using KPhone 3.xx and having no problems. However, I would appreciate it if someone more experienced could cast their eyes over this to make sure I'm not doing something stupid :) My Asterisk sip.conf is simply: [general] port = 5060 bindaddr = 0.0.0.0 context = default [myphone] type=friend host=dynamic context=default qualify=1000 And the KPhone configuration is set as follows: User part of SIP URL: myphone Host part of SIP URL: gdh Outbound Proxy: tel 'gdh' is the name of my local PC and resolves to 194.200.209.137 'tel' is the remote Asterisk server at 213.2.4.46. I have made sure there is no firewall issue between the two by two catch-all iptables -A FORWARD -s 194.200.209.137 -d 213.2.4.46 -j ACCEPT and iptables -A FORWARD -d 194.200.209.137 -s 213.2.4.46 -j ACCEPT at the firewall on each side. I've attached a gzipped snippet from ethereal in libpcap format of the failing registration. If the list doesn't permit attachments, it's also available at http://bum.net/sip.cap Cheers, Gavin. ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_oss.c:461: error: too many arguments to function `ast_queue_frame'
I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/07/04-11:28:50\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC-c -o chan_oss.o chan_oss.c chan_oss.c: In function `oss_call': chan_oss.c:461: error: too many arguments to function `ast_queue_frame' chan_oss.c:467: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `oss_new': chan_oss.c:712: warning: assignment from incompatible pointer type chan_oss.c: In function `console_answer': chan_oss.c:809: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_sendtext': chan_oss.c:841: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_hangup': chan_oss.c:861: error: too many arguments to function `ast_queue_hangup' chan_oss.c: In function `console_dial': chan_oss.c:883: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_transfer': chan_oss.c:935: error: too many arguments to function `ast_async_goto' make[1]: *** [chan_oss.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadBRI and UK ISDN2e
Morning Asterikians, I've just got my nice shiny quadBRI card, and it seems to be working very well - except for one little issue - CallerID. The card is currently connected to an ISDN2e line in P2P mode, and an S0 adapter on our existing alcatel PBX. The S0 connection recieves callerID and displays it correctly - the 2e line doesn't, and BT have said that CLID was enabled on the line two days ago. Does anyone have any pointers on this? My configuration is avaliable on request. Thanks, Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadBRI and UK ISDN2e
I've just got my nice shiny quadBRI card, and it seems to be working very well - except for one little issue - CallerID. The card is currently connected to an ISDN2e line in P2P mode, and an S0 adapter on our existing alcatel PBX. The S0 connection recieves callerID and displays it correctly - the 2e line doesn't, and BT have said that CLID was enabled on the line two days ago. Does anyone have any pointers on this? Just a quick check, is it connected to 'real' ISDN2e or a Business Highway ISDN port? If the later, make sure that BT have turned caller display on the ISDN port and not on one of the analogue ports - this is a common mistake they make when taking your order. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP flashhook transfer
Hello * users I try to get SIP flashhook transfer to work properly in my setup. The problem is that when I flashhook and then dial another extension I get some really garbled sound in the end I flashhook from. The remote can hear me just fine, I have threewaycalling=yes and transfer=yes in my sip.conf. Here is a complete overview of the sequence when I try to transfer the call: 1. Caller 1 dials some number with our PBX as destination 2. Employee 1 picks up the call 3. Employee 1 wants to transfer this call to Employee 2 at our company (But Employee 1 wants to talk to Employee 2 first, Supervised transfer) 4. Employee 1 flashhook his phone and Caller 1 gets Music-On-Hold 5. Employee 1 call Employee 2 and Employee 2 confirms that he can take the call. But now a problem arises: 6. When Employee 1 hangs up in order to let Caller 1 and Employee 2 have a conversation, Music-On-Hold start for Employee 2 also. In this state Caller 1 has Music-On-Hold forever or until he hangs up too. The 2 employees are in the same context in extension.conf, below is a snip of my sip.conf file: ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls disallow = all allow = ulaw threewaycalling = yes transfer = yes tos = 184 [43300634] type=friend secret= host=dynamic dtmfmode=inband defaultip=10.1.1.254 callerid=34 callgroup=1 pickupgroup=1 restrictcid=yes [43300645] type=friend username=43300645 secret= pickupgroup=1 callgroup=1 dtmfmode=inband host=dynamic defaultip=10.1.1.6 callerid=45 Regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadBRI and UK ISDN2e
Linus Surguy wrote: I've just got my nice shiny quadBRI card, and it seems to be working very well - except for one little issue - CallerID. The card is currently connected to an ISDN2e line in P2P mode, and an S0 adapter on our existing alcatel PBX. The S0 connection recieves callerID and displays it correctly - the 2e line doesn't, and BT have said that CLID was enabled on the line two days ago. Does anyone have any pointers on this? Just a quick check, is it connected to 'real' ISDN2e or a Business Highway ISDN port? If the later, make sure that BT have turned caller display on the ISDN port and not on one of the analogue ports - this is a common mistake they make when taking your order. It's a real ISDN2e circuit. I can't see why it's picking up the CallerID from the S0 adapter (which is apparantly configured in the same way as an ISDN2e from BT, according to our comms supplier) and not from the real 2e. Thanks, Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange SIP issue (again)
Hi, just to repeat my previous post (and trying to find a solution): Setup is * behind NAT. I can use FWD (time service, echo server) without problems when I add this to sip.conf: externip=a.b.c.d; a.b.c.d is the IP of the router (Linux/Nat) outside_addr=a.b.c.d My ICH however now responds with: -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 213.137.73.140 If I leave the above section out, ICH works fine for outbound calls but FWD is quiet. I have yet to get inbound calls to work. So fix this scenario, my suggestion would be to change the * sources so that the externip/outside_addr variables can be set/disabled for individual SIP providers rather than globally. Regards Andreas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail-hangup issue
Yes!, The latest CVS has fixed this problem. Thanks for the help. Sean - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, April 06, 2004 6:53 PM Subject: Re: [Asterisk-Users] voicemail-hangup issue I have a small * installation with 2 incomming analog lines connected to 2 X100P's, and several SIP phones. With the most recent update (cvs-04/01/04), I have started to see a problem. If someone is connected to voicemail, and hangs up without leaving a message (the problem does not occur if a message is left), asterisk does not register the hangup. The zaptel channel is then left open indefinatly. The end result is that both of the incomming analog lines become tied up, and callers get a busy signal. The problem does not seem to occur if the x100p caller hangs up at any other point durring a call. Does anyone know what this could be? Any help would be greatly appreciated. Others have commented about this same problem recently, and apparently its associated with a bug that was introduced in the zap channel. I'm running the same config with cvs from this morning, and I've not seen the problem today at all. (My previous update was about a month ago, and it wasn't present at that time.) As sort of a temp fix, you can add a parameter to the voicemail.conf to limit the voicemail duration before dumping the call. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange SIP issue (again)
Andreas, below is my partial sip.conf (which is relevant for fwd) this works for me. jakob [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to nat=yes ; externip = myhost.dyns.net ; Addr put in SIP messages if we're behind a NAT localnet = 192.168.10.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask ; ;* ; REGISTER WITH SIP PROVIDER ;* ; register = 54501:[EMAIL PROTECTED]/1234 ; Register with FWD as 1234 ;* ; OUTBOUND SIP CHANNELS ;* [fwd] type=friend secret=xxx username=54501 host=fwd.pulver.com ;outboundproxy=192.168.69.247:5082 ; not sure if this is implemented canreinvite=no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_motv: Request for Comment
excellent idea. eliot On Tue, 2004-04-06 at 23:31, Mark Spencer wrote: I've been considering the nature of Asterisk, its security, the bug tracker, and more... And i've come up with an interesting idea: A message of the version. The idea is that Asterisk has a compile time 32-bit unsigned int version which is incremented whenever some major new bug is fixed. When Asterisk starts up (and periodically, maybe once per day), it sends a packet with the version number to a server at Digium, along with a message level (INFO,MINOR,MAJOR,CRITICAL) and the Digium server replies (if it receives the packet, if not, it might get sent again in a day) with any INFO, MINOR, MAJOR, or CRITICAL messages which are associated with that version of the code. In this way, an asterisk administrator could easily see if there were any major issues, critical security updates, etc, that his system might need to be updated for. Now, of course, any time you put a call home feature in, there are people who will be concerned about privacy. Clearly it will be able to be disabled, but I want to run my idea about deployment by everyone here and see if you guys had some ideas. The idea would be that *new* installs (make samples) would have the feature turned on for MAJOR level by default, and that any existing install (e.g. /etc/asterisk/sip.conf exists, but not /etc/asterisk/motv.conf) would have the file created at the next make install based upon prompting the installer. Any feedback on: a) The idea itself -- is it a good one or is it stupid? b) The way to make it deployed without sneaking a call home in on anybody that doesn't want it? Thanks! Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: res_motv: Request for Comment
Mark Spencer [EMAIL PROTECTED] wrote: I've been considering the nature of Asterisk, its security, the bug tracker, and more... And i've come up with an interesting idea: A message of the version. The idea is that Asterisk has a compile time [...] a) The idea itself -- is it a good one or is it stupid? It could be a useful feature *if* done right. Some other people have already made some good comments on this. I think it seems like somewhat of a waste of time with the number of truly useful things that could be done to Asterisk instead. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_motv: Request for Comment
Mark Spencer wrote: Any feedback on: a) The idea itself -- is it a good one or is it stupid? Now this is just my views. No I do not feel we need to be sending any information back unless we want to. Like someone else said a sub job that is turned off by default. My preference would be no communication back. I would like to see on you web site more information on stable builds, bugs and easyer way to determine the version your running. Also maybe some feed back form that we can fill out and sumit to you. But all of them are manual and not automatic. b) The way to make it deployed without sneaking a call home in on anybody that doesn't want it? Thanks! Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The maximum capacity of MeetMe
Robert Hajime Lanning wrote: quote who="Andrew Thompson" I regret that I've only used MeetMe a few times, and only up to two users. Well, the problem with giving general stats, is that it REALLY depends on the exact environment. Key points: (on a server dedicated for conferences only) o number of channels o types of channels o codecs used (and ratio) o number of conferences o number of channels in the conferences Then givin the interupt load, cpu load, i/o load, memory load and bandwidth for each of these variables, you can find what hardware will run the load you want. If people are looking for a higher-capacity conferencing application, take a look at app_conference, in the iaxclient (on sourceforge) CVS. I haven't really benchmarked meetme, but I _think_ that app_conference might be able to beat it. Certainly in it's designed application it will (iax clients which use VAD on the client side).
Re: [Asterisk-Users] res_motv: Request for Comment
I wouldn't want a call home feature that is enabled by default. I think it would be great though if * had some ability to update itself. Perhaps via a CLI command, as others have suggested. Something similar to RH's up2date would be great in my opinion. Anyway, thats my 2 cents. Sean Rodger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with IAX2?
Andrew Kohlsmith wrote: Are there open problems/issues with iax2 and jitter (quality)? Just upgraded to today's dev cvs about an hour ago, and it seems the iax conversations are lower quality then a month or two ago. iax2 show firmware says version 13. (Test call originated from C7960 with g711.) I noticed the same thing. Jitter buffer apparently is broken, and has always been. I was advised to say jitterbuffer=no in iax.conf, but I swear it's better with it set to yes and then executing "iax2 set jitter 250" in the CLI. At least it was before I cvs up'd. :-) I found a jitter buffer bug in IAX2 a short while ago. It could potentially lead to misordered frames in conversations, and does so quite often when the sender of frames is using iaxclient under win9x. I compensated for this with a change in iaxclient, but the problem could also happen in asterisk-generated frames. See : http://sourceforge.net/mailarchive/forum.php?thread_id=4096021forum_id=29380 I don't know if this is the bug people are hitting, or not, though. Jeremy (of NuFone fame) has his jitterbuffer=no on his servers and since he's my VOIP provider I tend to just try and match his setup in terms of IAX2 anyway. I dunno, I do agree with you that it seemed better a while ago. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Siemens EWSD 13
Hi all, Has anyone got any experience with hooking Asterisk up with a Siemens EWSD 13 switch over a E1/PRI ? We're located in Belgium (Europe) and one of our telecom partners uses this switch. We connected one of our TE410P ports with their switch, but the status light on the TE410P card keeps blinking red. On their side they are getting a DSA (distance service alarm) error, so this normally means the devices 'see' eachother.. but there are still problems with the signalling. Our config below is the same as we are using for MCI, one of our other telecom partners. We tried changing the LBO and timing, but no luck. As you see the signalling is carried over channel 16 (default). TX and RX have also been regularly switched, so no luck.. Their switch is providing the timing. The telecom operator has double checked the asterisk config several times, and it's conform to their setup. The only thing they couldn't find in the Asterisk config is a 'multiframing' option. But I presume this is automatically detected or set by default ? They also tried normal/single(?) framing, but no difference. The card has also been tested with our MCI E1, and works flawlessly, so no hardware issue. Anyone got any further ideas ? Any info or help greatly appreciated! Our config, *** zaptel.conf *** span=1,1,6,ccs,hdb3,crc4,yellow bchan=1-15 bchan=17-31 dchan=16 *** zapata.conf *** [channels] switchtype=euroisdn signalling=pri_cpe pridialplan=unknown group=1 channel = 1-15,17-31 other zapata standard config ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with IAX2?
See : http://sourceforge.net/mailarchive/forum.php?thread_id=4096021forum_id=293 80 SourceForge reports invalid forum -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI-T400P-Asterisk-T400P-Adtran 750(L36 Firmware)-RAS Server. I have 4 Zap channels signalled FXO_KS to the 750 with FXS_LS channels, On-Hook messaging disabled, the rest defaults for the channels. In zapata.conf I've tried with both busydetect=yes and busydetect=no busycount=6, busycount=10, callprogress=yes, callprogress=no all combinations. The weird thing is, that if I forward the incoming call from the PRI out another channel on the PRI into a POTS line hooked into the RAS server, the connection is fine. In my view, that rules out the PRI and points the blame at either how the adtran is configured, or the how the channel itself is configured. Can anyone with a _working_ configuration similar to this chime in with some config info on the Zap channel and the channel bank config? Thanks in advance. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] res_motv: Request for Comment
Andy Powell Wrote: I'd like to give this one 10 thumbs down. IMHO a bad idea, a nasty little bad idea.. evil, spawn of Satan. If this were implemented the first job of a new update would be to rip it out and flush it down the nearest toilet. Just curious, but why does it strike you as such a bad idea? Especially if it was disabled by default. I can understand you not wanting your system security or your personal privacy compromised, but I think it would be great to have it in place for: A) Manual activation for those who want automated updates. B) CLI execution for occasional comparison to the current set. Perhaps it should be possible to flag the request with a token indicating that you don't want to be part of the survey, and you don't want your IP/host information stored. A --anon option, if you will. I can only wait until we see M$ like activation implemented... oh the joy... I am going to guess that you're joking. I just don't see that happening. Mark and the team at Digium seem dedicated to open source and to the Asterisk community. His asking for comments on this idea is a pretty good indicator of his concern for the community's opinions. It would be much better just to have the information present on either the Digium site or some other location. I see little point in wasting your valuable time doing something like this when there are so many outstanding issues and feature requests that could offer more. Perhaps Mark's time could be spent on other things, but I would still like to see Digium offer this option -- perhaps one of the other developers could head up the effort? Just putting the current version information up on a web page is nice, but it doesn't allow me to automatically query the system and discover known issues and possible solutions. I think that, for service providers that could very well be a critical next step. Several of my clients made the decision to go with RedHat based on their update service (for which they gladly paid). I would like to see the idea implemented, but I would like to see the above security considerations implemented (manual activation, a CLI option, and an anonymous option). I would also like to see the service extended into all of the Applications, Resources, Codecs, Formats, and other major components of Asterisk. It would be great to know when changes to both the core Asterisk engine and the associated modules are made, what the changes are, and also when new applications/resources/codecs, etc. are added to the CVS. Thoughts? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The maximum capacity of MeetMe
On Wed, 7 Apr 2004, Steve Kann wrote: If people are looking for a higher-capacity conferencing application, take a look at app_conference, in the iaxclient (on sourceforge) CVS. I haven't really benchmarked meetme, but I _think_ that app_conference might be able to beat it. Certainly in it's designed application it will (iax clients which use VAD on the client side). Which version of asterisk is it meant to compile against? It seems lite the calls to ast_set_read_format and ast_set_write_format are missing the needlock parameter. /Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
Bisker, Scott (7805) wrote: I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI-T400P-Asterisk-T400P-Adtran 750(L36 Firmware)-RAS Server. I have 4 Zap channels signalled FXO_KS to the 750 with FXS_LS channels, On-Hook messaging disabled, the rest defaults for the channels. In zapata.conf I've tried with both busydetect=yes and busydetect=no busycount=6, busycount=10, callprogress=yes, callprogress=no all combinations. We have 4 750's and one TSU 600 working with PC anywhere for data communications for our support department. We have on this system 2 T400P's. The only thing I can say is who are you getting your timing from. We are able to get modem calls and faxes without problems. But this is only using PRI from Allegenice. We also have a LD service T1 from Sprint that is in no way able to handle any data calls. Our Adtrans are out of the box without any changes to them. This is our settings in our zapata.conf. ; Enable echo cancellation echocancel=yes ;echocancelwhenbridged=yes immediate=no ;adsi=yes usecallerid=yes hidecallerid=no ;callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes musiconhold=default signalling = fxo_ks Hope this helps. The weird thing is, that if I forward the incoming call from the PRI out another channel on the PRI into a POTS line hooked into the RAS server, the connection is fine. In my view, that rules out the PRI and points the blame at either how the adtran is configured, or the how the channel itself is configured. Can anyone with a _working_ configuration similar to this chime in with some config info on the Zap channel and the channel bank config? Thanks in advance. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTel toll-free gateway
Title: IAXTel toll-free gateway Is anyone else having trouble placing toll-free calls though IAXTel lately? Mine just stopped working yesterday, yet I seem to be able to make 1-700 calls. -brian 1-700-676-3830
[Asterisk-Users] Dial Capi Question / Problem
Is it possible to detect the attempt to dial to unallocated (unassigned)
numbers ? Currently I cannot distinguish the error from no-answer.
There is a extension with priority n + 101 but it is not used.
The dialplan extension looks like:
Dial(CAPI/${num1}:B${num2},30,T)
If I use lowercase b (always early B3) I can hear some external
error indication for 30 seconds.
${CAUSECODE} and ${HANGUPCAUSE} do not contain usefull information.
--
Stefan Tichy [EMAIL PROTECTED]
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[Asterisk-Users] Out of trunk data space on call number 16386, dropping
Hi all, We keep getting these and all the calls between these two asterisk boxes get dropped. what is going on here, I have been trying to solve this problem on my own but maybe I don't have the trunk setup right. also I have posed the output of my full log of the machine with the zap interface, the other is using ztdummy. IAX.conf on machine 1: [general] port=5036 ;iaxcompat=yes bandwidth=low disallow=ilbc disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=ulaw ;allow=gsm ; Always allow GSM, it's cool :) jitterbuffer=no trunkfreq=20 ;dropcount=3 ;maxjitterbuffer=500 ;maxexcessbuffer=100 ; tos=lowdelay register = [EMAIL PROTECTED] register = [EMAIL PROTECTED] ; [woodlane] allow=ulaw ;allow=gsm type=friend jitterbuffer=no username=woodlane context=dialout host=dynamic trunk=yes trunkfreq=20 IAX.conf on machine2: [general] port=5036 bindaddr = XXX.XXX.XXX.XXX iaxcompat=yes ;amaflags=default ;accountcode=lss0101 bandwidth=low disallow=ilbc disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=ulaw disallow=gsm; Always allow GSM, it's cool :) jitterbuffer=no ;dropcount=3 ;maxjitterbuffer=500 maxexcessbuffer=100 trunkfreq=20; How frequently to send trunk msgs (in ms) register = [EMAIL PROTECTED] authdebug=yes tos=lowdelay [lachnet] allow=ulaw disallow=ilbc disallow=lpc10 disallow=gsm jitterbuffer=no username=lachnet type=friend trunk=yes trunkfreq=20 host=dynamic ;secret=telco context=default include = dialout [woodlane] allow=ulaw ;allow=gsm type=friend jitterbuffer=no username=woodlane context=dialout host=dynamic trunk=yes trunkfreq=20 Full.log: Apr 7 09:41:21 DEBUG[704531]: Bridge stops bridging channels [EMAIL PROTECTED]/16385 and Zap/1-1 Apr 7 09:41:21 DEBUG[704531]: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Apr 7 09:41:21 DEBUG[704531]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Apr 7 09:41:21 DEBUG[704531]: disabled echo cancellation on channel 1 Apr 7 09:41:21 DEBUG[704531]: Set option TDD MODE, value: OFF(0) on Zap/1-1 Apr 7 09:41:21 DEBUG[704531]: Updated conferencing on 1, with 0 conference users Apr 7 09:41:21 DEBUG[704531]: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Apr 7 09:41:21 DEBUG[704531]: disabled echo cancellation on channel 1 Apr 7 09:41:21 VERBOSE[704531]: -- Hungup 'Zap/1-1' Apr 7 09:41:21 VERBOSE[704531]: == Spawn extension (dialout, 5522307, 1) exited non-zero on '[EMAIL PROTECTED]/16385' Apr 7 09:41:21 DEBUG[704531]: We're hanging up [EMAIL PROTECTED]/16385 now... Apr 7 09:41:21 VERBOSE[704531]: -- Hungup '[EMAIL PROTECTED]/16385' Apr 7 09:41:29 DEBUG[163851]: Made call 5 into trunk call 16386 Apr 7 09:41:29 VERBOSE[163851]: -- Accepting unauthenticated call from 65.113.15.19, requested format = 4, actual format = 4 Apr 7 09:41:29 VERBOSE[737299]: -- Executing Dial([EMAIL PROTECTED]/16386, Zap/g1/BYEXTENSION) in new stack Apr 7 09:41:29 VERBOSE[737299]: -- Called g1/5522307 Apr 7 09:41:29 DEBUG[163851]: Ooh, voice format changed to 4 Apr 7 09:41:30 DEBUG[114696]: Enabled echo cancellation on channel 1 Apr 7 09:41:30 VERBOSE[737299]: -- Zap/1-1 is ringing Apr 7 09:41:35 DEBUG[114696]: Echo cancellation already on Apr 7 09:41:35 VERBOSE[737299]: -- Zap/1-1 answered [EMAIL PROTECTED]/16386 Apr 7 09:41:35 WARNING[737299]: Out of trunk data space on call number 16386, dropping Apr 7 09:41:44 DEBUG[163851]: Made call 8 into trunk call 16387 Apr 7 09:41:44 VERBOSE[163851]: -- Accepting unauthenticated call from 65.113.15.19, requested format = 4, actual format = 4 Apr 7 09:41:44 VERBOSE[753684]: -- Executing Dial([EMAIL PROTECTED]/16387, Zap/g1/BYEXTENSION) in new stack Apr 7 09:41:44 VERBOSE[753684]: -- Called g1/5540408 Apr 7 09:41:44 DEBUG[163851]: Ooh, voice format changed to 4 Apr 7 09:41:46 DEBUG[114696]: Enabled echo cancellation on channel 2 Apr 7 09:41:46 VERBOSE[753684]: -- Zap/2-1 is ringing Apr 7 09:41:49 DEBUG[114696]: Echo cancellation already on Apr 7 09:41:49 VERBOSE[753684]: -- Zap/2-1 answered [EMAIL PROTECTED]/16387 Apr 7 09:42:04 VERBOSE[114696]: -- Channel 1, span 1 got hangup Apr 7 09:42:04 DEBUG[737299]: Bridge stops because we're zombie or need a soft hangup: [EMAIL PROTECTED]/16386, c1=Zap/1-1, flags: No,No,No,Yes Apr 7 09:42:04 DEBUG[737299]: Bridge stops bridging channels [EMAIL PROTECTED]/16386 and Zap/1-1 Apr 7 09:42:04 DEBUG[737299]: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Apr 7 09:42:04 DEBUG[737299]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Apr 7 09:42:04 DEBUG[737299]: disabled echo cancellation on channel 1 Apr 7 09:42:04 DEBUG[737299]: Set option TDD MODE, value: OFF(0) on Zap/1-1 Apr 7 09:42:04 DEBUG[737299]: Updated conferencing on 1, with 0 conference users Apr 7 09:42:04 DEBUG[737299]: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Apr 7 09:42:04
Re: [Asterisk-Users] Callerid + Zaphfc
Hi, use prilocaldialplan=local in zapata.conf. -- best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mi, 2004-04-07 um 09.59 schrieb Martin Schenkelberg: Hi all, i have an ISDN Phone connected to an HFC-S based card, all works fine but is i call the Phone from a SIP User Agent or over PSTN Line the Phones Display shows the correct CallerID but with a leading 0 . I cant find this in the config files, how can is solve this? Dialing Out with the ISDN Phone transmitts the correct callerid. Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAPRTC question(s)
I have a system with no Digium hardware in it (two others with 2 X100P cards in each of them as well). I'm interested in using MeetMe in the one without the hardware (it works great in the ones with the hardware). I can't use ztdummy, because the system has usb-ohci drivers, rather than usb-uhci. I have read the little there is about zaprtc, and I am wondering whether there is a downside in turning off RTC support in the kernel, and recompiling. Are there other things that might break if I do this (it simply feels like a more drastic step than the ztdummy approach)? (I am running Red Hat 9.0) Finally, and this will show my complete naivete for linux programming, I am curious as to why no one has written a timer that simply hooks the standard kernel installed RTC? From the rtc.txt file in the Documentation directory of the kernel source, it seems that one can hook the interrupt and get the clock ticks delivered via interrupt directly to your c code. Isn't that what is needed to get a stable timing device in *? Just curious, as I'm sure that it's way more sophisticated than that... Thanks in advance. P.S. The system with no Digium hardware in it is in a colo facility that is 250 miles from my house, and besides, I don't have physical access to the machine. So, it would be painful, and expensive, for me to arrange for a Digium card to be installed in the machine, and it would be used for nothing other than the clock, since there are no other interfaces available for me to plug into the card. This was just to nip the why don't you just pony up for a Digium card? responses :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no good day today ! :(
Today I updated my asterisk cvs, I would love to set up mysql support for voiceboxes. Here is what has happened (by now): cvs does not compile as downloaded, module chan_oss reports an error; if I compile with -i all works fine except for chan_oss. Ok, time to set USE_MYSQL_VM_INTERFACE=1 in apps\Makefile and to do make install -i When I launch asterisk, (with chan_oss excluded)I get a broken pipe message repeated until crash, disabling music on hold solves the problem. Compiling again with USE_MYSQL_VM_INTERFACE=0 in apps\Makefile solves at least the broken pipe problem. I'm banging my head on the wall from this morning .. is it possible that the cvs I downloaded has something to fix ? tnx for any help ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTel toll-free gateway
Is anyone else having trouble placing toll-free calls though IAXTel lately? Mine just stopped working yesterday, yet I seem to be able to make 1-700 calls. It's up/down/etc rather frequently, so no surprise. Good thing it's not a paid service or we'd all have an issue. Consider it as a temporary testing facility, not a production resource. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank?
Four or five analog lines can be done with a single computer so no channel bank is needed. If you need 6 or more than there is also the choice of using two machines and IAX. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Hajime Lanning Sent: Tuesday, April 06, 2004 12:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channel Bank? quote who=Ken Hello, I'm new to Asterisk and would like to know how you could have 4 to 6 incoming analog POTS lines connecting to the Asterisk server and have 4 to 6 analog lines going out.(A T1 line is too costly). Would 2 channel banks be used? A T1 channelbank has 24 channels, so only 1 is needed. FXO channels (What you connect to the POTS lines) can be both inbound and outbound. If you are not using DID. So, you just need to find out how many concurrent calls you need to support. If you are using analog DID lines, then those are inbound only, and require FXS ports. (You supply dialtone and battery, the carrier's switch picks up your line and dials into your PBX.) Now, there are multiple ways to get the analog lines into Asterisk... o use an external gateway... POTS - SIP - Asterisk o wait until next month and get the FXO multiport cards from Digium o get a T1 card + channelbank -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
Same as mine. Do you know off the top of your head what firwmare you're using? Also, what RAS card do you have on your PCAnywhere side? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista Sent: Wednesday, April 07, 2004 10:39 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750 Bisker, Scott (7805) wrote: I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI-T400P-Asterisk-T400P-Adtran 750(L36 Firmware)-RAS Server. I have 4 Zap channels signalled FXO_KS to the 750 with FXS_LS channels, On-Hook messaging disabled, the rest defaults for the channels. In zapata.conf I've tried with both busydetect=yes and busydetect=no busycount=6, busycount=10, callprogress=yes, callprogress=no all combinations. We have 4 750's and one TSU 600 working with PC anywhere for data communications for our support department. We have on this system 2 T400P's. The only thing I can say is who are you getting your timing from. We are able to get modem calls and faxes without problems. But this is only using PRI from Allegenice. We also have a LD service T1 from Sprint that is in no way able to handle any data calls. Our Adtrans are out of the box without any changes to them. This is our settings in our zapata.conf. ; Enable echo cancellation echocancel=yes ;echocancelwhenbridged=yes immediate=no ;adsi=yes usecallerid=yes hidecallerid=no ;callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes musiconhold=default signalling = fxo_ks Hope this helps. The weird thing is, that if I forward the incoming call from the PRI out another channel on the PRI into a POTS line hooked into the RAS server, the connection is fine. In my view, that rules out the PRI and points the blame at either how the adtran is configured, or the how the channel itself is configured. Can anyone with a _working_ configuration similar to this chime in with some config info on the Zap channel and the channel bank config? Thanks in advance. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_motv: Request for Comment
On Tue, 2004-04-06 at 23:31, Mark Spencer wrote: Any feedback on: a) The idea itself -- is it a good one or is it stupid? I like the idea of being able to see what updates/fixes are available vs. the code that I'm running. I think this would definitely be helpful to me. b) The way to make it deployed without sneaking a call home in on anybody that doesn't want it? Like others on here, I'd like to see it as a console command - I'd like to be able to come into the office in the morning and type 'check motv' on the console, and see if there's anything I need/want. Having it auto-phone-home on startup wouldn't be too useful for me, since this would only occur when we were already performing an upgrade. As far as the periodic message, again that wouldn't be too useful, I wouldn't be looking at the console to see the results. jwsh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_motv: Request for Comment
Hi. another (stupid) thing. don't call that function motv. motv is a name for another opensource project. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
Bisker, Scott (7805) wrote: Same as mine. Do you know off the top of your head what firwmare you're using? Also, what RAS card do you have on your PCAnywhere side? I have firmware L36. Ras card is a Digikey 4 port board on one NT server and others are using the normal serial ports on the servers. The desktops are using there modems connected to there PC's via Serial cables. All our modems are USR Sporters 56K we have about 20 of them. Except for 3 USR Courier 56K. For our fax board we are using BrookTrout I4P on a Windows 2000 server with ZataFax. Everything is working off the timing from the PRI line. Asterisk is older on this installation. This installation is still using .5 from CVS 12/05/03. I belive if it works leave it along! And it works just fine! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadBRI and UK ISDN2e
Which brand of card did you get? Sincerely, Stephen Karrington Dreamtime.net Inc. http://www.dreamtime.net http://www.emailblaster.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation. ===8==Original message text=== Morning Asterikians, I've just got my nice shiny quadBRI card, and it seems to be working very well - except for one little issue - CallerID. The card is currently connected to an ISDN2e line in P2P mode, and an S0 adapter on our existing alcatel PBX. The S0 connection recieves callerID and displays it correctly - the 2e line doesn't, and BT have said that CLID was enabled on the line two days ago. Does anyone have any pointers on this? My configuration is avaliable on request. Thanks, Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ===8===End of original message text=== ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting info about changes in CVS
There are several ways to know what changes in Asterisk's CVS. This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly up to date CVS changelog summary information. You can also sign up for the Asterisk-CVS mailing list at http://lists.digium.com/mailman/listinfo/asterisk-cvs Archives of the Asterisk-CVS mailing list are at http://lists.digium.com/pipermail/asterisk-cvs/ -- Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the Unofficial Links) and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Proxy Problem (NAT Environment)
Title: Message Then your firewall is closing the return RTPport to fast. Check for the latest firmware and also make sure the SPI (stateful packet inspection) is turned off if you router has that option. Otherwise you may have to give up and fall back to port forwarding. Michael Shuler, C.E.O.BitWise Systems, Inc.1301 W. Pioneer ParkwayPeoria, IL 61615Office: (217) 585-0357Cell: (309) 657-6365Fax: (309) 213-3500E-Mail: [EMAIL PROTECTED]Customer Service: (877) 976-0711 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Markus MiertschinkSent: Tuesday, April 06, 2004 10:58 AMTo: [EMAIL PROTECTED]Subject: AW: [Asterisk-Users] SIP Proxy Problem (NAT Environment) It is set to yes Strangely it works only if I make the call from one direction one voice channel gets no voice transmitted Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Michael ShulerGesendet: Dienstag, 6. April 2004 17:30An: [EMAIL PROTECTED]Betreff: RE: [Asterisk-Users] SIP Proxy Problem (NAT Environment) Make sure you have nat=yes for the sip.conf entry for SIPGATE. Michael Shuler, C.E.O.BitWise Systems, Inc.1301 W. Pioneer ParkwayPeoria, IL 61615Office: (217) 585-0357Cell: (309) 657-6365Fax: (309) 213-3500E-Mail: [EMAIL PROTECTED]Customer Service: (877) 976-0711 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Markus MiertschinkSent: Tuesday, April 06, 2004 10:09 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] SIP Proxy Problem (NAT Environment) I have the problem calling from a NAT firewalled SIP Phone to sipgate.de. Somehow it is working. The connection works. Everything seems to be fine. I could talk to the external phone. But: the external voice gets not transmitted to my internal (calling) phone. Calling inbound from SipGate to the same phone works perfectly. Both channels are perfectly proxied. The * is used as a proxy and registered the sipgate account. This way works: SIPGATE ---NAT(IP0)Asterisk(IP1)-IP-Phone(IP2) This not (partially): SIPGATE--NAT(IP0)---Asterisk(IP1)---IP-Phone(IP2) All ports are mapped to the address of IP1. If I look into the tcpdump log it seems that all ports used there are matching my NAT settings. Asterisk is happy. No problems. I only run into immediate abortion of the call if Asterisk is configured using outbound_address with the ip of IP0. I have no clue what to do anymore Regards, Markus Virus checked by G DATA AntiVirusKitVersion: AVK 14.0.635 from 31.03.2004Virus news: www.antiviruslab.com
RE: [Asterisk-Users] Channel Bank?
quote who=John Vogel Four or five analog lines can be done with a single computer so no channel bank is needed. If you need 6 or more than there is also the choice of using two machines and IAX. Talk about port density issues. So, if he really needs all 12 lines, then he needs 3 PCs? (He probably doesn't need all 12.) -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another software like Asterisk?
Hi! I am looking for a software that can work as h.323 - sip gateway other than asterisk and free. Someone can help me? Thanks. Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ANNOUNCEMENT : MeetMe Web User Interface
Hello Asterimaniacs, Finally, I went out with that... sorry I had lot of work and not enough courage to work at night ;) Well, mysql and postgresql now work well for me and I have put some order in the code. Just enjoy it, I m waiting for the feedbacks ;) http://www.areski.net/asterisk-meetme/about.php Disclaimer : Use at your own risk ! To remember: The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has been changed. Kind regards, Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belad Arezqui URL : www.areski.net Tl. : (+34) 650 78 43 55 E-mail : [EMAIL PROTECTED] [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Caller ID and Voicemail ?s
Wow... talk about a detailed response; thanks! In our situation, we've got a T-1 voice PRI from Allegiance Telcom. For the benefit of those of us who aren't as in the know as you are (and who have no affiliation with a CLEC), is there a way to be able to control what gets sent out as our name portion of the Caller ID (even if it means changing what's recorded at Allegiance)? We somehow manage to do so with the number part. In other words, type real slow and mention specific conf files if possible. This is pretty new stuff for me... Thanks again! -- Ryan On Apr 6, 2004, at 7:59 PM, Kyle Thomas wrote: SCP=Service control point (database that houses name to number) SCP DIP = Query to an SCP via the SS7 network ISUP = SS7 signaling for call setup and teardown (equivalent of invite,ringing,ok,bye) IAM = Initial address message (equal to the SIP invite ) LNP= Local number portability (uses the SS7 network as a backbone). This let's people keep thier phone number and switch service providers. There is nothing quick about quick caller id. The far end Telco will override the name infomration sent to the PSTN and perform thier dips regardless, overwriting the info you are trying sending out. We are a CLEC so, therefore we store, therefore it works.. On Tue, 6 Apr 2004, Andrew Kohlsmith wrote: The terminating telco is doing an SCP dip to thier local SCP's and the database probably does not have that name mapped to this number. First thing to do is make sure the generic name ISUP optional paramter is set in the outgoing IAM / ISDN setup from your GW. You could also store with an SS7 provider , if these are ported numbers you are sending out make sure that the CNAM field in the LNP line record is set to the point code alias of the provider you are storing with. The terminating switch will first do an LNP dip to see what CNAM alias to launch the CNAM dip to. If that is not found , will default to the local SCP thus not finding your record. Ok, and now for the rest of us... SCP? SCP dip? ISUP? IAM? LNP? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John Vogel wrote: | Four or five analog lines can be done with a single computer so no channel | bank is needed. If you need 6 or more than there is also the choice of using | two machines and IAX. I assume you would be using 4 or 5 X101Ps (or similar). The only problem I have currently with our systems that are connected via IAX2 switching is echo. We have a single incoming line to one machine with an X101P in it. We then have another machine with a TDM400P and a single analog extension. The machines are connected via IAX2. Just today, one of our clients complained that he heard himself echoing. Usually we hear ourselves echoing but the person on the other end does not hear themselves echoing in the configuration we have setup. Haven't tried putting both cards into the same computer yet, but that will take a lot of fiddling, since the new machine has an nForce2 chipset and likes to assign the same IRQ to lots of different things. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAdC8euYsUrHkpYtARAic3AJsGQ7bRzlBh3MFG/SgZlFm7cL3+0QCeNs0S meELq3WfHKjVuKN640RBwCg= =Ra2J -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] B-channels resetting every 60 minutes?
Hello, As you can see are pri is being reset every 60 minutes! Is there a way to stop this?? Is it a Zapata configuration problem? We have a * box with a single port T1/pri card installed. Thanks lach Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 4 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 5 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 6 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 7 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 8 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 9 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 10 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 11 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 12 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 13 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 14 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 15 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 16 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 17 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 18 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 19 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 20 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 21 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 22 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 23 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 4 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 5 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 6 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 7 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 8 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 9 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 10 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 11 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 12 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 13 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 14 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 15 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 16 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 17 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 18 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 19 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 20 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 21 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 22 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 23 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 1 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 4 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 5 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 6 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 7 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 8 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 9 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 10 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 11 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 12 successfully restarted on span 1
Re: [Asterisk-Users] Another software like Asterisk?
Try Vovida's Vocal, i think it does it. Mireia Munoz de jesus wrote: Hi! I am looking for a software that can work as h.323 - sip gateway other than asterisk and free. Someone can help me? Thanks. Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Struggling with ISDN4Linux and Asterisk config
The card is an ASUSCOM ISDNLink PCI (passive) and the circuit is from Qwest (in the US). I will be using this circuit only for voice (I'm doing this because of the poor quality of my POTS lines). I've compiled Hisax (as a module) into my 2.4.25 kernel, and with 'modprobe hisax type=35 protocol=4 id=hisax' I get the following: Apr 7 10:34:24 dev kernel: HiSax: Linux Driver for passive ISDN cards Apr 7 10:34:24 dev kernel: HiSax: Version 3.5 (module) Apr 7 10:34:24 dev kernel: HiSax: Layer1 Revision 1.1.4.1 Apr 7 10:34:24 dev kernel: HiSax: Layer2 Revision 1.1.4.1 Apr 7 10:34:24 dev kernel: HiSax: TeiMgr Revision 1.1.4.1 Apr 7 10:34:24 dev kernel: HiSax: Layer3 Revision 1.1.4.1 Apr 7 10:34:24 dev kernel: HiSax: LinkLayer Revision 1.1.4.1 Apr 7 10:34:24 dev kernel: HiSax: Approval certification failed because of Apr 7 10:34:24 dev kernel: HiSax: unauthorized source code changes Apr 7 10:34:24 dev kernel: HiSax: Card 1 Protocol NI1 Id=hisax (0) Apr 7 10:34:24 dev kernel: HiSax: HFC-PCI driver Rev. 1.1.4.1 Apr 7 10:34:24 dev kernel: PCI: Enabling device 00:0a.0 ( - 0003) Apr 7 10:34:24 dev kernel: HiSax: HFC-PCI card manufacturer: Asuscom/Askey card name: 675 Apr 7 10:34:24 dev kernel: HFC-PCI: defined at mem 0xe0a2 fifo 0xd6178000(0x16178000) IRQ 11 HZ 100 Apr 7 10:34:24 dev kernel: HFC_PCI: resetting card Apr 7 10:34:24 dev kernel: HFC 2BDS0 PCI: IRQ 11 count 226490 Apr 7 10:34:25 dev kernel: HFC 2BDS0 PCI: IRQ 11 count 226524 Apr 7 10:34:25 dev kernel: HiSax: National ISDN-1 Rev. 1.1.4.1 Apr 7 10:34:25 dev kernel: HiSax: National ISDN-1 Rev. 1.1.4.1 Apr 7 10:34:25 dev kernel: HiSax: 2 channels added Apr 7 10:34:25 dev kernel: HiSax: MAX_WAITING_CALLS added From this I assume the card is correctly configured (I don't know anything about the unauthorized source code changes). In modem.conf I have: [interfaces] driver=i4l language=en type=autodetect dialtype=tone mode=immediate context = isdn group = 4 msn= incomingmsn = 6791578 device = /dev/ttyI0 incomingmsn = 6791608 device = /dev/ttyI1 I don't know where the SPID should go. In MSN? In extensions.conf I have: [isdn] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,Goto(mainmenu|s|4) [mainmenu] exten = s,1,Wait(1) exten = fax,1,Goto(fax|1|1); print the fax exten = s,2,Answer exten = s,3,Wait(1) exten = s,4,SetMusicOnHold(default) exten = s,5,DigitTimeout,5 exten = s,6,ResponseTimeout,10 exten = s,7,Background(introspect/welcome) exten = s,8,Background(introspect/dialextension) exten = s,9,Background(silence/2) exten = s,10,Dial(Zap/10Zap/7,20,t) ... etc. When I dial into the PBX I get the following from Asterisk: -- Executing Wait(Modem[i4l]/ttyI0, 1) in new stack -- Executing Answer(Modem[i4l]/ttyI0, ) in new stack == Spawn extension (isdn, s, 2) exited non-zero on 'Modem[i4l]/ttyI0'e to answer: (No Response) -- Hungup 'Modem[i4l]/ttyI0' and the following in the logs: Apr 7 10:22:16 dev kernel: isdn_net: call from 3036388531 - 0 6791578 ignored Apr 7 10:22:16 dev kernel: isdn_tty: call from 3036388531, - RING on ttyI0 Apr 7 10:22:16 dev kernel: isdn_net: call from 3036388531 - 0 6791578 ignored Apr 7 10:22:16 dev kernel: isdn_tty: call from 3036388531 - 6791578 ignored Apr 7 10:22:18 dev kernel: isdn_net: call from 3036388531 - 0 6791578 ignored Apr 7 10:22:18 dev kernel: isdn_tty: call from 3036388531 - 6791578 ignored Apr 7 10:22:59 dev kernel: isdn: hisax,ch1 cause: 0066 Sometimes I also get: Apr 7 10:59:51 dev kernel: SPID not supplied in EAZMSN When I dial out from the PBX I get the following from Asterisk: -- Starting simple switch on 'Zap/7-1' -- Executing Dial(Zap/7-1, Modem/g4/3036741234) in new stack Apr 7 10:49:02 WARNING[311316]: chan_modem.c:181 modem_call: Destination g4/3036740068 requres a real destination (device:destination) -- Couldn't call g4/3036740068 -- Hungup 'Modem[i4l]/ttyI1' == Everyone is busy at this time -- Executing Congestion(Zap/7-1, ) in new stack == Spawn extension (local, 93036741234, 2) exited non-zero on 'Zap/7-1' -- Hungup 'Zap/7-1' and nothing in the logs. Can someone help me with this? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens EWSD 13
Hi, If you didnt do it yet I'd suggest you start with simpliest thing which is making loop on the cables. And testing the status. Simple RJ45 plug with 10cm of 2 pairs of cable crimped by 1,2 on 4,5. When you make loop T410P goes green. EWSD also will see a loop. Check cables and loops and when you are sure that if you do loop just before EWSD T410P goes green and when you do loop just bofore T410P the EWSD see it. Then plug it directly and use zttool to make software loopback on T410P which EWSD should be capable to see. Then if its ok run ztcfg -vvv check if modules are loaded and then run asterisk and read debug messages. Try to run pri debug span x to see if there is any info or even intense debug. But I'd bet on the cables. and chcking both sides of the cable Anyone got any further ideas ? Any info or help greatly appreciated! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk / SMP / Scalability
I've got Asterisk loading 100,000+ extensions in extensions.conf. This process is taking a little upwards of 10 minutes to complete on each of my dual 3.2Ghz HP DL380 with SuSE Linux Enterprise 8 boxes. Although asterisk creates child processes, it appears that it is only using a single processor to parse extensions.conf. I've turned off Hyper Threading on the servers which has increased the extensions.conf parsing speed, but not by more than a couple minutes. Is this a bug, or simply the way Asterisk works during startup? If it is the way Asterisk works during startup, would it be safe to say that once started - that the child processes would function? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: pda skype
I don't have a PocketPC PDA, mine is Palm. But regardless, I don't see what all the Hype is with Skype. It is a closed protocol and highly platform-restricted product. Sure the concept of a peer-to-peer phone network is interesting, but if not everyone can connect to it, what is the point? If they want to keep certain features under their control so that they can eventually charge for it, then by all means do so. But if they would release the basic protocol specs, so that others can access the network in general, they would see many more users. I for one am not going to run yet another soft phone and/or IM client on my system just to connect to yet another phone network. A friend of mine and I tried it when it first came out, and it worked about as well as FWD, or IAXTel, or Firefly, or... You get the point. Now if I could attach my Asterisk server to it and be able to make and receive simple voice calls with other users, that would be great. I don't need hotlist functionality, if I dial their number and they aren't on, I get a busy reorder signal. No big deal. They definitely have a good idea, in the fact that it works, doesn't have too many problems with firewalls, and is not server reliant. But keeping it closed is preventing a lot of people from joining them. -Original Message- From: Jason A. Pattie [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 07, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FW: pda skype -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dean Collins wrote: | | | http://www.skype.net/download_pda.html Hmm. I don't have a 400MHz PDA running PocketPC. I only have a 206MHz PDA (running Familiar Linux and GPE). - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAdC3fuYsUrHkpYtARAjn8AJ0dslvk0TWf/RSjN11246XkWOH35QCfcuBR HC2I8QTK3F2zyST3Ayz6G8s= =X0GT -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Getting info about changes in CVS
Eric Wieling wrote: There are several ways to know what changes in Asterisk's CVS. snip You can also sign up for the Asterisk-CVS mailing list at http://lists.digium.com/mailman/listinfo/asterisk-cvs I've signed up for the cvs mailing list and have been stockpiling the messages. I planned on building something that would read them, and parse them into a database for viewing on the web, but I just haven't had time yet. If there is interest, I'll put down the newborn and go back to hacking... - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Caller ID and Voicemail ?s
I cannot help you with the .conf files on the * as I am brand new to the * and in the process of compiling the software now. I do know this.. You have to make sure the the generic name IE (information element ) is being populated in the outbound ISDN setup message to Allegiance. If you have a protocol analyzer you could check this , or maybe Allegiance an check this for you. At any rate, they should get this parameter inbound from you ISDN and then pass this parameter to the PSTN (most likely via ISUP). I would get with Allegiance and make sure they are setup to pass CNAM to the PSTN with you. Or you can tell them what ANI/CPN you are sending and they probably store with an SS7 provider , and they can update the SCP for the desired name that you want for any ANI. There might be a cost for this... This last option is the cleanest way to do it. I currently do this for customer's that have PRI PBX's and sit on my switch with a T-1 Kyle On Wed, 2004-04-07 at 12:30, Ryan Thrash wrote: Wow... talk about a detailed response; thanks! In our situation, we've got a T-1 voice PRI from Allegiance Telcom. For the benefit of those of us who aren't as in the know as you are (and who have no affiliation with a CLEC), is there a way to be able to control what gets sent out as our name portion of the Caller ID (even if it means changing what's recorded at Allegiance)? We somehow manage to do so with the number part. In other words, type real slow and mention specific conf files if possible. This is pretty new stuff for me... Thanks again! -- Ryan On Apr 6, 2004, at 7:59 PM, Kyle Thomas wrote: SCP=Service control point (database that houses name to number) SCP DIP = Query to an SCP via the SS7 network ISUP = SS7 signaling for call setup and teardown (equivalent of invite,ringing,ok,bye) IAM = Initial address message (equal to the SIP invite ) LNP= Local number portability (uses the SS7 network as a backbone). This let's people keep thier phone number and switch service providers. There is nothing quick about quick caller id. The far end Telco will override the name infomration sent to the PSTN and perform thier dips regardless, overwriting the info you are trying sending out. We are a CLEC so, therefore we store, therefore it works.. On Tue, 6 Apr 2004, Andrew Kohlsmith wrote: The terminating telco is doing an SCP dip to thier local SCP's and the database probably does not have that name mapped to this number. First thing to do is make sure the generic name ISUP optional paramter is set in the outgoing IAM / ISDN setup from your GW. You could also store with an SS7 provider , if these are ported numbers you are sending out make sure that the CNAM field in the LNP line record is set to the point code alias of the provider you are storing with. The terminating switch will first do an LNP dip to see what CNAM alias to launch the CNAM dip to. If that is not found , will default to the local SCP thus not finding your record. Ok, and now for the rest of us... SCP? SCP dip? ISUP? IAM? LNP? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kyle Thomas Director of Engineering Monmouth Telephone 732-704-1000 x 130 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lucent Phones
Does Asterisk work with Lucent or any other PBX phone systems ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] B-channels resetting every 60 minutes?
Hello Lach- I think B channel restarting is a normal occurrence, although I thought it was supposed to be more often than once every 60 minutes. The channels are not supposed to be restarted if they show in use, so this should be a transparent occurrence. Are there any problems that this is causing? Cheers Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of osx Sent: Wednesday, April 07, 2004 5:47 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] B-channels resetting every 60 minutes? Hello, As you can see are pri is being reset every 60 minutes! Is there a way to stop this?? Is it a Zapata configuration problem? We have a * box with a single port T1/pri card installed. Thanks lach Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 4 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 5 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 6 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 7 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 8 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 9 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 10 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 11 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 12 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 13 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 14 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 15 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 16 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 17 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 18 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 19 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 20 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 21 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 22 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 23 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 4 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 5 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 6 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 7 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 8 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 9 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 10 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 11 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 12 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 13 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 14 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 15 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 16 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 17 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 18 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 19 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 20 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 21 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 22 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 23 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 1 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 3 successfully restarted on span
[Asterisk-Users] Call hangs up after a fiew seconds with a quad BRI
Hi All Just got a new quadBRI card and connected one port to our Old PBX. When I make a call from a sip phone to a phone number the phone rings, I hook up, and the call on the sip phone allmost imidialely disconnects, after a fiew seconds the real phone disconnects too. Here is a trace: -- Executing SetCallerID(SIP/cramer1-b718, 45) in new stack -- Executing SetCIDName(SIP/cramer1-b718, 45) in new stack -- Executing Dial(SIP/cramer1-b718, Zap/g1/00796085427) in new stack -- Called g1/00796085427 Apr 7 19:38:14 WARNING[114696]: chan_zap.c:6009 zt_pri_error: PRI: received TEI check request for TEI = 127 Apr 7 19:38:15 WARNING[114696]: chan_zap.c:6009 zt_pri_error: PRI: received TEI check request for TEI = 127 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/cramer1-b718 Apr 7 19:38:21 WARNING[114696]: chan_zap.c:4032 zt_new: Channel 1 already has a Real call -- Executing Dial(Zap/1-1, Zap/g2/00796085427|60|tTr) in new stack Apr 7 19:38:21 NOTICE[344086]: app_dial.c:536 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time -- Accepting call from '' to '00796085427' on channel 1, span 1 Apr 7 19:38:21 WARNING[327701]: chan_zap.c:602 zt_get_index: Unable to get index, and nullok is not asserted Apr 7 19:38:21 WARNING[327701]: chan_zap.c:3772 zt_write: Zap/1-1 doesn't really exist? -- Hungup 'Zap/1-1' == Spawn extension (sip, , 3) exited non-zero on 'SIP/cramer1-b718' Apr 7 19:38:31 WARNING[344086]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in context 'sip' -- Hungup 'Zap/1-1' extension.conf: exten = ,1,SetCallerID(45) exten = ,2,SetCIDName(45) exten = ,3,Dial(Zap/g1/00796085427) zapata.conf: [channels] ; ; ISDN quadBRI interfaces ; switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local usecallerid=yes callerid=0448474545 group = 1 context=sip channel = 1-2 switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local group = 2 context=sip channel = 4-5 switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local group = 3 context=sip channel = 7-8 switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local group = 4 context=sip channel = 10-11 Any Ideas ? Best regards Matthias -- _;\_Matthias Cramer / mc322-ripe System Network Manager /_. \ Dolphins Network Systems AGPhone +41-44-847'45'45 |/ -\ .) Libernstrasse 24 Fax +41-44-847'45'49 -'^`- \; CH-8112 Otelfingen http://www.dolphins.ch/ GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 pgp0.pgp Description: PGP signature
[Asterisk-Users] callback with 3 way call?
i'm new with asterisk. i currently have 1 fxo port. my phone line connected to the fxo is capable of 3 way calls using flash. i'm thinking of a callback and then 3 way call from asterisk. is this possible? 1. phone1 calls asterisk thru zap/fxo. asterisk gets callerid of phone1. 2. asterisk will callback phone1 using zap/fxo. 3. phone1 answers and is prompted for a number to call. 4. phone1 dials a number. 5. asterisks intiates a flash on zap/fxo. dials the number and then flash again zap/fxo for 3way call. i saw callback scripts but haven't found a sample script to do a flash and dial the number. i tried doing callback, flash, dial and flash but didn't work. tried callback, flash, senddtmf, and flash and didn't work either. any tips? thanks. __ Do you Yahoo!? Yahoo! Small Business $15K Web Design Giveaway http://promotions.yahoo.com/design_giveaway/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] B-channels resetting every 60 minutes?
osx wrote: Hello, As you can see are pri is being reset every 60 minutes! Is there a way to stop this?? Is it a Zapata configuration problem? We have a * box with a single port T1/pri card installed. This is an expected and desired behavior. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] errror compiling asterisk from cvs
I am getting this too under RH9. Sean - Original Message - From: Alessio Focardi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 07, 2004 9:04 AM Subject: [Asterisk-Users] errror compiling asterisk from cvs I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/07/04-11:28:50\ -DINSTA LL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk \ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPO OLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPAT H=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_ PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC-c -o chan_oss.o chan_oss.c chan_oss.c: In function `oss_call': chan_oss.c:461: error: too many arguments to function `ast_queue_frame' chan_oss.c:467: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `oss_new': chan_oss.c:712: warning: assignment from incompatible pointer type chan_oss.c: In function `console_answer': chan_oss.c:809: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_sendtext': chan_oss.c:841: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_hangup': chan_oss.c:861: error: too many arguments to function `ast_queue_hangup' chan_oss.c: In function `console_dial': chan_oss.c:883: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_transfer': chan_oss.c:935: error: too many arguments to function `ast_async_goto' make[1]: *** [chan_oss.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
Title: RE: [Asterisk-Users] Lucent Phones Absolutely, it can be a little tricky but its definitely doable. Check out the info I wrote on the wiki, as well as peoples posts here for more information on hows its done. Matt -Original Message- From: James Moran [mailto:[EMAIL PROTECTED]] Sent: Wednesday, April 07, 2004 1:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Lucent Phones Does Asterisk work with Lucent or any other PBX phone systems ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
I'm timing off my PRI from Verizon as well. This is mind boggling. All my Fax machines are fine. The modems connect, but drop the calls after about 1-2 minutes regardless of busydetect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista Sent: Wednesday, April 07, 2004 12:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750 Bisker, Scott (7805) wrote: Same as mine. Do you know off the top of your head what firwmare you're using? Also, what RAS card do you have on your PCAnywhere side? I have firmware L36. Ras card is a Digikey 4 port board on one NT server and others are using the normal serial ports on the servers. The desktops are using there modems connected to there PC's via Serial cables. All our modems are USR Sporters 56K we have about 20 of them. Except for 3 USR Courier 56K. For our fax board we are using BrookTrout I4P on a Windows 2000 server with ZataFax. Everything is working off the timing from the PRI line. Asterisk is older on this installation. This installation is still using .5 from CVS 12/05/03. I belive if it works leave it along! And it works just fine! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- Subject: [Asterisk-Users] Lucent Phones Does Asterisk work with Lucent or any other PBX phone systems Sure. You can use Asterisk as a VoIP gateway to your existing legacy PBX. You can't plug Lucent's (Avaya's) DCP, MLX, or ATL phone sets into an Asterisk box -- the protocols are all proprietary. But you can certainly connect between the systems using analog or T1/Ei connections. Regards, Steve Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] B-channels resetting every 60 minutes?
Hi Lach, this looks like normal behaviour to me. Most of the equipment I use issues a restart upon initial physical connection (bad equipment can cause problems when it doesn't do this) and then several times per hour thereafter. Once every hour seems infrequent but I guess that this is down to individual suppliers' interpretation of the specification documents. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of osx Sent: 07 April 2004 17:47 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] B-channels resetting every 60 minutes? Hello, As you can see are pri is being reset every 60 minutes! Is there a way to stop this?? Is it a Zapata configuration problem? We have a * box with a single port T1/pri card installed. Thanks lach Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 4 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 5 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 6 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 7 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 8 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 9 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 10 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 11 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 12 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 13 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 14 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 15 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 16 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 17 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 18 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 19 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 20 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 21 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 22 successfully restarted on span 1 Apr 7 09:00:07 VERBOSE[114696]: -- B-channel 23 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 4 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 5 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 6 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 7 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 8 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 9 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 10 successfully restarted on span 1 Apr 7 10:00:08 VERBOSE[114696]: -- B-channel 11 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 12 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 13 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 14 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 15 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 16 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 17 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 18 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 19 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 20 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 21 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 22 successfully restarted on span 1 Apr 7 10:00:09 VERBOSE[114696]: -- B-channel 23 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 1 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 2 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 3 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 4 successfully restarted on span 1 Apr 7 11:00:10 VERBOSE[114696]: -- B-channel 5
[Asterisk-Users] Re: res_motv: Request for comment
One thing that the BSD open source operating system projects do, and many other projects for that matter, which Asterisk does not seem to do, is put CVS ID tags in the source files of the package itself. If ID tags were put into the source files, and even embedded in strings so that theyshowed up in the binary files too, that would go a long way toward helping users determine which version of Asterisk they had, and where they were relative to the current state of the development tree. It seems like this change requires no real coding, just adding a line or two to each source file, and CVS does the rest for you by bumping the version numbers as changes come in. Another advantage of this approach, is that users can succinctly and accurately point out which versions of which modules work and which ones contain critical bugs. Then you can say things like: File res_moh.c, V1.25 and later contains the fix you're looking for. Just a thought. -Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk / SMP / Scalability
quote who=Darren Sessions I've got Asterisk loading 100,000+ extensions in extensions.conf. This process is taking a little upwards of 10 minutes to complete on each of my dual 3.2Ghz HP DL380 with SuSE Linux Enterprise 8 boxes. Although asterisk creates child processes, it appears that it is only using a single processor to parse extensions.conf. I've turned off Hyper Threading on the servers which has increased the extensions.conf parsing speed, but not by more than a couple minutes. Is this a bug, or simply the way Asterisk works during startup? If it is the way Asterisk works during startup, would it be safe to say that once started - that the child processes would function? This behaviour is just for parsing *.conf files. You may want to put the extensions into a database and use an AGI script to perform extension routing. (Though, I think it would bypass CDR.) -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange SIP issue (again)
My FWD and ICH through NAT work just fine (for outgoing calls) depending on the setup I choose. The setup is just mutually exclusive. FWD needs: externip=a.b.c.d; a.b.c.d is the IP of the router (Linux/Nat) outside_addr=a.b.c.d; as per your config this is optional ICH doesn't need this though - in fact it rejects the call with an error if the above lines are set. Obviously proxy recovery is implemented differently for the two providers. I'd like to get both services working together and I think it would be best if * allows for more fine grained control of the SIP messages related to NAT. This might already be implemented and I just don't know how to use it ... On Wed, 2004-04-07 at 08:55, Jakob Strebel wrote: Andreas, below is my partial sip.conf (which is relevant for fwd) this works for me. jakob [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to nat=yes ; externip = myhost.dyns.net ; Addr put in SIP messages if we're behind a NAT localnet = 192.168.10.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask ; ;* ; REGISTER WITH SIP PROVIDER ;* ; register = 54501:[EMAIL PROTECTED]/1234 ; Register with FWD as 1234 ;* ; OUTBOUND SIP CHANNELS ;* [fwd] type=friend secret=xxx username=54501 host=fwd.pulver.com ;outboundproxy=192.168.69.247:5082 ; not sure if this is implemented canreinvite=no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750
Bisker, Scott (7805) wrote: I'm timing off my PRI from Verizon as well. This is mind boggling. All my Fax machines are fine. The modems connect, but drop the calls after about 1-2 minutes regardless of busydetect. That was our exact problem with Sprint when we had there T1 line. We decided to switch to Allegence and problem was gone. I just tired a data connection through the Sprint LD line we have and after 2 minutes we got dropped. If we tell the modem to only use lower speed like 28.8 it will stay connected for longer time. At 19.2 no problem they will stay connnected. Verizon might have the same problem with there lines like Sprint. That there not data lines but voice only. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista Sent: Wednesday, April 07, 2004 12:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config. Adtran 750 Bisker, Scott (7805) wrote: Same as mine. Do you know off the top of your head what firwmare you're using? Also, what RAS card do you have on your PCAnywhere side? I have firmware L36. Ras card is a Digikey 4 port board on one NT server and others are using the normal serial ports on the servers. The desktops are using there modems connected to there PC's via Serial cables. All our modems are USR Sporters 56K we have about 20 of them. Except for 3 USR Courier 56K. For our fax board we are using BrookTrout I4P on a Windows 2000 server with ZataFax. Everything is working off the timing from the PRI line. Asterisk is older on this installation. This installation is still using .5 from CVS 12/05/03. I belive if it works leave it along! And it works just fine! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAXTel toll-free gateway
Brian == Brian Cuthie [EMAIL PROTECTED] writes: Brian Is anyone else having trouble placing toll-free calls though Brian IAXTel lately? Mine just stopped working yesterday, yet I Brian seem to be able to make 1-700 calls. I'd suggest using enum lookups on freenum.org instead. Cf: http://www.mail-archive.com/[EMAIL PROTECTED]/msg23732.html -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oss.c:461: error: too many arguments to function `ast_queue_frame'
Alessio Focardi wrote: I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/07/04-11:28:50\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC-c -o chan_oss.o chan_oss.c chan_oss.c: In function `oss_call': chan_oss.c:461: error: too many arguments to function `ast_queue_frame' chan_oss.c:467: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `oss_new': chan_oss.c:712: warning: assignment from incompatible pointer type chan_oss.c: In function `console_answer': chan_oss.c:809: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_sendtext': chan_oss.c:841: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_hangup': chan_oss.c:861: error: too many arguments to function `ast_queue_hangup' chan_oss.c: In function `console_dial': chan_oss.c:883: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `console_transfer': chan_oss.c:935: error: too many arguments to function `ast_async_goto' make[1]: *** [chan_oss.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 It appears that the final argument to all these functions (normally a 0 or 1) has been dropped, but it hasn't been fixed in chan_oss.c or chan_alsa.c. If you happen to have already compiled asterisk before and aren't doing a clean recompile then it appears that the problem isn't spotted and recompiled (poor dependency checking in the Makefile?) The easy fix is just to drop the final arguments for all these functions and then to kick off the compile again. Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] B-channels resetting every 60 minutes?
We have one other error (twice today) we get Out of trunk data space on call number , dropping How do I determine what is causing this error? we have a point-to-point T1 between 2 * boxes, with 3 phone in the remote office. I have no idea how the trunk could be out of space. The end goal is trying to figure why we are dropping calls! Are you running IAX between the boxes? Are you running IAX trunking between the boxes? If so.. Do you have trunking configured identically on both ends (trunk=yes in iax.conf)? Do you have a zaptel device (or ztdummy) in both ends? If either of those questions is no, then you'll get the out of trunk data space error and drop calls. Make sure both ends are configured the same for trunking and you have a zaptel device or ztdummy. Or just don't use trunking (trunk=no). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Getting info about changes in CVS
Eric Wieling wrote: There are several ways to know what changes in Asterisk's CVS. This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly up to date CVS changelog summary information. You can also sign up for the Asterisk-CVS mailing list at http://lists.digium.com/mailman/listinfo/asterisk-cvs Archives of the Asterisk-CVS mailing list are at http://lists.digium.com/pipermail/asterisk-cvs/ Is there any reason that CVSView is not installed and publically viewable? It might help the who don't know all the CVS CLI commands get a graphical (and colored) view of the lines added and changed over time? Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk call manager
I am trying to setup the call manager and I configured the manager.conf file. When I try to telnet 0.0.0.0 5038 It says trying 0.0.0.0 Connected to localhost Escape character is '^]'. Asterisk Call Manager/1.0 Then I type Action:Login (enter) Username:sam Secret:sam Then I enter twice I get Response: error Message: missing action in request I am not sure what it means. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Out of trunk data space on call number 16386, dropping
I'm having the same kind of issues. We get the out of trunk data space error consistently during conference calls between asterisk servers. And occasionally on regular iax calls. Also while we're on a conference call it seems to cause other calls going out through iax to fail and also give this error. (weather its to another asterisk server or through say oneunified) If you figure this out, please let us know here. I'm pretty much at a loss as to what could be causing it. Justin Carlson wrote: Hi all, We keep getting these and all the calls between these two asterisk boxes get dropped. what is going on here, I have been trying to solve this problem on my own but maybe I don't have the trunk setup right. also I have posed the output of my full log of the machine with the zap interface, the other is using ztdummy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] attendent transfer on ZAP channels
Hi Il mer, 2004-04-07 alle 20:28, Bartosz Jozwiak ha scritto: hello, Is it possible to make attendant transfer (not blind) with ZAP channels ? sure. just press the flash key on the phone (also known as the 'R' key, at least in EU), you will hear the dialtone, while the caller is put on hold. dial the extension you wanna transfer to, speak with the remote party and then: hangup to transfer to the dialled exten OR press R to be in a 3-way conference (of course the remote party should not hangup) OR just press R to get the call back (and the remote party should hangup) OR press R twice to get the call back is the remote party doesn't hangup immediately threewaycall and transfer must be enabled into zapata.conf matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk call manager
Hi. try adding a whitespace between ':' and the command. Eg. action: login enter blah blah Matteo. I am trying to setup the call manager and I configured the manager.conf file. When I try to telnet 0.0.0.0 5038 It says trying 0.0.0.0 Connected to localhost Escape character is '^]'. Asterisk Call Manager/1.0 Then I type Action:Login (enter) Username:sam Secret:sam Then I enter twice I get Response: error Message: missing action in request I am not sure what it means. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmgi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] B-channels resetting every 60 minutes?
I'd like to jump in here because we're also experiencing the out of trunk data problem. So is this the only thing that causes the out of trunk data error? Because we are running iax between the boxes and both boxes have trunk=yes in the iax.conf entries and there is a zaptel device in both. James Sharp wrote: Are you running IAX between the boxes? Are you running IAX trunking between the boxes? If so.. Do you have trunking configured identically on both ends (trunk=yes in iax.conf)? Do you have a zaptel device (or ztdummy) in both ends? If either of those questions is no, then you'll get the out of trunk data space error and drop calls. Make sure both ends are configured the same for trunking and you have a zaptel device or ztdummy. Or just don't use trunking (trunk=no). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with IAX2?
Hi I am also having jitter trouble on IAX2, and I can vouch that the jitter buffer is busted. On Wed, 07 Apr 2004 09:56:01 -0400 Steve Kann [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: Are there open problems/issues with iax2 and jitter (quality)? Just upgraded to today's dev cvs about an hour ago, and it seems the iax conversations are lower quality then a month or two ago. iax2 show firmware says version 13. (Test call originated from C7960 with g711.) I noticed the same thing. Jitter buffer apparently is broken, and has always been. I was advised to say jitterbuffer=no in iax.conf, but I swear it's better with it set to yes and then executing iax2 set jitter 250 in the CLI. At least it was before I cvs up'd. :-) I found a jitter buffer bug in IAX2 a short while ago. It could potentially lead to misordered frames in conversations, and does so quite often when the sender of frames is using iaxclient under win9x. I compensated for this with a change in iaxclient, but the problem could also happen in asterisk-generated frames. See : http://sourceforge.net/mailarchive/forum.php?thread_id=4096021forum_id=29380 I don't know if this is the bug people are hitting, or not, though. Jeremy (of NuFone fame) has his jitterbuffer=no on his servers and since he's my VOIP provider I tend to just try and match his setup in terms of IAX2 anyway. I dunno, I do agree with you that it seemed better a while ago. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Herbalife Independent Distributor http://www.healthiest.co.za ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk dimensioning (IVR, mass calling)
Hi, I presently have 6 PRIs of IVR traffic that I am planning to migrate from Dialogic on SCO-Unix to Digium-Asterisk on Debian. Here is the general description of the traffic in question : - IVR system, 138 PRI channels (6 PRIs, multiple D-channel) - Some traffic from TV ads, so all traffic typically arrives within a few seconds - Up to 12-15 simul. outbound (external) conferencing calls for customer service - No internal phones - Several HTTPS transactions per minute - MySQL queries (to separate server) - Want do drop minimum of calls, mostly 900 services I am assuming the following things, performance-wise : - Since I will be recording to and playing back from ?-law, which is the format used by PRI in North America, I will require no codec translation and that should be the easiest on the CPU (right?). - I will need no echo cancellation since I am exclusively on PRI (right?). If I rely on previous posts, mainly from Scott Stingel and Azher Amin, I won't be able to put all this in a single server. I would instead have 2-3 servers (ex. Xeon bi-proc.) with 2-3 PRIs each, and NFS for dynamic content (recordings from callers). Does this sound realistic? Is it too risky to go with Asterisk with such a set-up? Thanks a lot for your help, Yves Chouinard Vox-Tel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk call manager
On Wed, 7 Apr 2004, Jain, Sonal wrote: I am trying to setup the call manager and I configured the manager.conf file. When I try to telnet 0.0.0.0 5038 It says trying 0.0.0.0 Connected to localhost Escape character is '^]'. Asterisk Call Manager/1.0 Then I type Action:Login (enter) Username:sam Secret:sam Then I enter twice I get Response: error Message: missing action in request I am not sure what it means. Thanks You need a space after each header. Action: Login Username: sam Secret: sam from doc/manager.txt: Command Syntax -- Management communication consists of tags of the form header: value, terminated with an empty newline (\r\n) in the style of SMTP, HTTP, and other headers. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
What about the Partner phones and TDM400? You can't plug Lucent's (Avaya's) DCP, MLX, or ATL phone sets into an Asterisk box -- the protocols are all proprietary. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Toshiba Digital Phones - Asterisk
I am planning an * install at my business. It will be replacing an existing Toshiba system (I think it is a 424dk). I was wondering if anyone knows of a way for me to use my existing Toshiba phones to connect to *. I would rather not have to spend the $15,000 to replace all of my phones, but I can't find any other way to do it. Your help is greatly appreciated. Thanks, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with IAX2?
dido -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 07, 2004 2:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with IAX2? Hi I am also having jitter trouble on IAX2, and I can vouch that the jitter buffer is busted. On Wed, 07 Apr 2004 09:56:01 -0400 Steve Kann [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: Are there open problems/issues with iax2 and jitter (quality)? Just upgraded to today's dev cvs about an hour ago, and it seems the iax conversations are lower quality then a month or two ago. iax2 show firmware says version 13. (Test call originated from C7960 with g711.) I noticed the same thing. Jitter buffer apparently is broken, and has always been. I was advised to say jitterbuffer=no in iax.conf, but I swear it's better with it set to yes and then executing iax2 set jitter 250 in the CLI. At least it was before I cvs up'd. :-) I found a jitter buffer bug in IAX2 a short while ago. It could potentially lead to misordered frames in conversations, and does so quite often when the sender of frames is using iaxclient under win9x. I compensated for this with a change in iaxclient, but the problem could also happen in asterisk-generated frames. See : http://sourceforge.net/mailarchive/forum.php?thread_id=4096021forum_id=2938 0 I don't know if this is the bug people are hitting, or not, though. Jeremy (of NuFone fame) has his jitterbuffer=no on his servers and since he's my VOIP provider I tend to just try and match his setup in terms of IAX2 anyway. I dunno, I do agree with you that it seemed better a while ago. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Herbalife Independent Distributor http://www.healthiest.co.za ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error 488 - Not Acceptable Here
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
are registered by the below information
*CLI sip show peers
Name/usernameHost Mask Port Status
2002/2002192.168.22.199 (D) 255.255.255.255 5060 Unmonitored
2001/2001192.168.22.200 (D) 255.255.255.255 5060 Unmonitored
2000/2000192.168.22.198 (D) 255.255.255.255 5060 Unmonitored
*CLI sip show users
Username Secret Authen Def.Context A/C
2002 ciscomd5,plaintextdemo No
2001 ciscomd5,plaintextdemo No
2000 ciscomd5,plaintextdemo No
All 3 phones and the asterisk box are on the 192.168.22.0/24 subnet.
I've attached my sip.conf and extensions.conf file for review...
When I start the server and a phone dials another phone I get the below
answer.
*CLI -- Executing Dial(SIP/2001-0bb5, SIP/2002|30|tr) in new stack
-- Called 2002
-- Got SIP response 488 Not Acceptable Here back from 192.168.22.199
== No one is available to answer at this time
-- Timeout on SIP/2001-0bb5
I *believe* the sip response might be from the phone itself - and not a
asterisk misconfig. I'm just wanting a second pair of eyes.
I put in
canreinvite=no
for each phone profile as people have said this is needed for buggy
Cisco phones.
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
;bindaddr = 192.168.22.254; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
tos = lowdelay; can be lowdelay, throughput, reliability, mincost
[2000]
type=friend ; This device takes and makes calls
username=2000 ; Username on device
secret=cisco; Password for device
;host=192.168.22.1 ; This host is not on the same IP addr every time
host=dynamic
context=demo; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
canreinvite=no
; voicemailbox has messages in it
[2001]; Duplicate of 2000, except with different auth data
type=friend
username=2001
secret=cisco
host=dynamic
;host=192.168.22.2
context=demo
mailbox=101
canreinvite=no
[2002]; Duplicate of 2000, except with different auth data
type=friend
username=2002
secret=cisco
;host=192.168.22.3
host=dynamic
context=demo
mailbox=102
canreinvite=no
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
[incoming]
exten = s,1,Echo ;for testing the connection
;exten = s,1,Playback,demo-thanks ;for playing a file
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without the ';')
; Note that this is different from the include command that includes contexts within
; other contexts. The #include command works in all asterisk configuration files.
;#include filename.conf
; The Globals category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[Asterisk-Users] Newbie question
Hey All, We are using Asterisks as a voicemail only application, and so far all is great. (Excellent product!) However, I do have one question that I am hoping you might be able to help me with. In our asterisk application. When our customers call *55 (our dialplan code to check voicemail) then they are sent directly to voicemail (asterisk). Asterisk then gives a voice prompt asking the customer to enter their extension number (entire 10 digit telephone number in our case). My question is. Is there a way to make asterisk aware of the calling-from (callerID) number so that it will automatically detect the number and then go directly to asking them to input their password. If so, where would I make the config changes for this in the asterisk config files, and does anyone have an example of a similar config? Thanks! Darren Nay VOIP Network Developer Ionosphere, Inc [EMAIL PROTECTED]
Re: [Asterisk-Users] dropped calls from queue
I just updated to latest cvs and the problem remains. I did also notice that when the call coming in on the queue is through a Zap line (from an adtran 750 to an x100p) it logs the following just before the warnings below: pr 7 14:21:21 VERBOSE[60194841]: -- SIP/hrutter-432b answered Zap/13-1 Apr 7 14:21:21 DEBUG[60194841]: Set option TONE VERIFY, mode: MUTECONF/MAX(2) on Zap/13-1 Apr 7 14:21:21 VERBOSE[60194841]: -- Stopped music on hold on Zap/13-1 Tony Buser wrote: We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it seems to happen when the following gets logged: Apr 7 14:53:35 WARNING[60424217]: File 10 does not exist in any format Apr 7 14:53:35 WARNING[60424217]: Unable to open 10 (format G729A): No such file or directory Apr 7 14:53:35 WARNING[60424217]: Agent on SIP/hrutter-c6fa hungup on the customer. They're going to be pissed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialpad.com
Greetings, Does anyone have any experience in getting dialpad.com working with * They use a proprietary softphone but also have facility for cisco ata-186 and Sipura SPA-2000. Before I go off and investigate, I though I would check and see if anyone has any experience with them Thanks, Craig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: pda skype
I'm going to leave most of what you said alone, I understand you point and it's your point to make. However I will make a small comment about I don't need hotlist functionality, if I dial their number and they aren't on, I get a busy reorder signal. No big deal. Presence based information is the biggest seller in the IP PBX market at the moment, being able to tell what/where a person is certainly driving a lot of sales through my door. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Hall Sent: Thursday, 8 April 2004 3:29 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: pda skype I don't have a PocketPC PDA, mine is Palm. But regardless, I don't see what all the Hype is with Skype. It is a closed protocol and highly platform-restricted product. Sure the concept of a peer-to-peer phone network is interesting, but if not everyone can connect to it, what is the point? If they want to keep certain features under their control so that they can eventually charge for it, then by all means do so. But if they would release the basic protocol specs, so that others can access the network in general, they would see many more users. I for one am not going to run yet another soft phone and/or IM client on my system just to connect to yet another phone network. A friend of mine and I tried it when it first came out, and it worked about as well as FWD, or IAXTel, or Firefly, or... You get the point. Now if I could attach my Asterisk server to it and be able to make and receive simple voice calls with other users, that would be great. I don't need hotlist functionality, if I dial their number and they aren't on, I get a busy reorder signal. No big deal. They definitely have a good idea, in the fact that it works, doesn't have too many problems with firewalls, and is not server reliant. But keeping it closed is preventing a lot of people from joining them.
Re: [Asterisk-Users] Getting info about changes in CVS
On Wed, 2004-04-07 at 17:20, Eric Wieling wrote: There are several ways to know what changes in Asterisk's CVS. This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly up to date CVS changelog summary information. You can also sign up for the Asterisk-CVS mailing list at http://lists.digium.com/mailman/listinfo/asterisk-cvs Archives of the Asterisk-CVS mailing list are at http://lists.digium.com/pipermail/asterisk-cvs/ Any chance of adding this list to the GMane archive? For me browsing list archives via NNTP is *much* nicer than web interfaces... Thanks a lot, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
