[asterisk-users] enum
Dear Please how can I make a local dns naptr on my system ,ro resolve local calls using enum Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Page + ParkAndAnnounce
[Sorry it's the third time I send this message as I couldn't see it in the list. I hope it will not come three times]. Hi everybody. It is possible to announce the parking position through a paging to a group of extensions? I would like that when someone parks a call, some phones will announce with the speaker the position. Something like: exten = s,1,ParkAndAnnounce(call-parked-at:PARKED|30|PAGE(LOCAL/[EMAIL PROTECTED] pageLOCAL/[EMAIL PROTECTED]|) Is there a way, maybe with a different approach? Thanks, Pol Po ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Show agent queue status on the phone?
I've not used the Cisco kit for this, but you might try adding 'hints' to your agent extensions, and then defining a BLF button to subscribe to this. e.g. If you have an agent with ID 1001, add this to extensions.conf (or equivalent)... exten = 1001,hint,Agent/1001 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Trumbull Sent: 14 December 2006 18:04 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Show agent queue status on the phone? Hi All, Is it possible to show an agent's queue status on the phone? For example, in our current non-asterisk PBX, if a member of a call queue does not answer the phone when a queue call is sent to them, they go to a 'not ready' status, and this is indicated on their phone. So when they return to their desk, they can see that they are not ready, so they hit a button to put themselves back into a ready status. I can accomplish the 'not ready' functionality by using the PauseQueueMember function, but now I need to somehow display the pause/unpause status on the phone so the staff member knows if they got paused. Does anybody know how to do this? This might be phone specific, so I'll mention that we are using Cisco 7961G's. Thanks -Kevin Trumbull ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Selecting outbound trunks
Hi Folks, Can you point me towards some info on how to specify that certain extensions use specific outbound trunks - we can only set outbound caller ID against the SIP accounts managed by our service provider (we cannot pass CID info to them at the moment although they are promising this facility) and so I'd like to do the following: Extn A + Extn B + Extn C - Outbound via SIP account 1 only Extn D - Outbound via SIP account 2 only I presume there's some grouping arrangement that will do this? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware TDM Switching
Is this only possible in a hard configuration way? or is it possible that asterisk handles the call and tell the zap channel, now you have to connect this 2 zap channels cross connect to each other? Thanks Nico PS: Sorry, but there is NO info about this on voip-info.org On Thu, 14 Dec 2006, Eric ManxPower Wieling wrote: [EMAIL PROTECTED] wrote: Do anybody know, if there is a way to connect 2 zap-channels with Hardware TDM Switching? It's called DACS. See the /etc/zapata.conf config file sample. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] enum
use dundi Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Khaled wrote: Dear Please how can I make a local dns naptr on my system ,ro resolve local calls using enum Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0658-1, 12/14/2006 - 12/15/2006 4:11:15 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to know who hangup ?
Hi, Using AMI or dial plan, how can i know which leg (channel ?) of a bridged call, hangup ? AMI send 2 hangup events, which have both cause 16 (normal clearing), and the first hangup event is the called leg hangup event, not the one who hangup. Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth.com on asterisk
They provided DIDs too. It was not that straight forward, but I've figured it out how to use them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call from h323 to SIP
Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten = 3298,2,Dial(SIP/[EMAIL PROTECTED]) If a make a call to callamanager CISCO that forward to 3298 i read in asterisk console: Log: Verbosity is at least 20 -- Executing Answer(H323/ip$172.z.z.z:4836/14, ) in new stack -- Executing Dial(H323/ip$172.z.z.z:4836/14, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/[EMAIL PROTECTED] is ringing Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to ulaw Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec ... translation path from g729 to slin Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to ulaw Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to slin Dec 15 14:45:13 WARNING[19794]: translate.c:116 ast_translator_build_path: No translator path from alaw to unknown Dec 15 14:45:13 WARNING[19794]: channel.c:1202 queue_frame_to_spies: Cannot build a path from g729 to slin Dec 15 14:45:13 WARNING[19794]: chan_h323.c:614 oh323_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 4/64) Dec 15 14:45:13 WARNING[19794]: chan_sip.c:2572 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Dec 15 14:45:13 WARNING[19794]: translate.c:116 ast_translator_build_path: No translator path from alaw to unknown Dec 15 14:45:13 WARNING[19794]: channel.c:2752 ast_channel_make_compatible: No path to translate from H323/ip$172.z.z.z:4836/14(256) to SIP/193.x.x.x-40455d68(8) Dec 15 14:45:13 WARNING[19794]: app_dial.c:1602 dial_exec_full: Had to drop call because I couldn't make H323/ip$172.z.z.z:4836/14 compatible with SIP/193.x.x.x-40455d68 == Spawn extension (default, 3298, 2) exited non-zero on 'H323/ip$172.z.z.z:4836/14' Why? where am i wrong? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is it possible to use Asterisk voicemail as anouncement system only?
Hello, we are using asterisk in combination with the voicemail system. I´m just wondering if it is possible to switch the voicemail to an I am on holiday mode. This means that the unavailable message is played to the caller but no possability to record a message. So far I did not find an option in the voicemail.conf for this. Any ideas except creating my own ivr menu ? best regards Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hardware TDM Switching
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, December 14, 2006 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hardware TDM Switching [EMAIL PROTECTED] wrote: Do anybody know, if there is a way to connect 2 zap-channels with Hardware TDM Switching? It's called DACS. See the /etc/zapata.conf config file sample. He means /etc/zaptel.conf I thinkright? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended Transfer on queue_log
I'm using asterisk blind/attended transfer feature on a queue (also tried with sip phones feature), and both type of transfers work fine. The problem is that attended trasfers doesn't get logged to queue_log, but blind transfers are logged just fine. Anyone knows if this is the correct behavior? -- Regards, Miguel Paolino ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hardware TDM Switching
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, December 14, 2006 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hardware TDM Switching [EMAIL PROTECTED] wrote: Do anybody know, if there is a way to connect 2 zap-channels with Hardware TDM Switching? It's called DACS. See the /etc/zapata.conf config file sample. Is there a way to do this dynamically? Something in the dialplan that would trigger this? I have calls coming in on one PRI and depending on the DID they go out on a second PRI (going to a dialup pool). I had hoped the Zaptel drivers would do a bridge of these channels but that doesn't happen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page + ParkAndAnnounce
On Friday 15 December 2006 4:18 am, Apesys wrote: exten = s,1,ParkAndAnnounce(call-parked-at:PARKED|30|PAGE(LOCAL/[EMAIL PROTECTED] o pageLOCAL/[EMAIL PROTECTED]|) why not Local/[EMAIL PROTECTED], and then have something like this: [group_page] exten = ,1,Dial(SIP/555) exten = ,1,Dial(SIP/123SIP/456SIP/789) exten = ,1,Dial(SIP/123SIP/789) ... is that closer to what you're looking for? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco Call Manager 4.0 to Asterisk, Anyone have SIP Reinvite working?
Hi All, I haven't started sip traces or debug yet, but was wondering what the deal is with the CCM and reinvite, why it doesn't work with Asterisk (using 1.2.9.1). I can make calls back and forth all day with canreinvite=no, when I try to reinvite an inbound sip call from the CCM with Asterisk server 1 to Asterisk Server 2, I get one-way audio issues. All the RTP ports are configured the same. I remember Cisco Phones and ATA's have some reinvite issues, wondering if the same applies to the CCM and how it handles reinvites? In the wiki example of integrating CCM with Asterisk, the SIP context shows canreinvite=yes, so should this be working ok, maybe I'm doing something wrong? Thanks JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 100rel Prack enable
Dear all, I'm trying to receive a call from a VoIP provider to my Asterisk which is behind a router, with a port forwarding (5060). This configuration has already been validated with another VoIP provider, but in the present case, not. I suppose (thanks to the sip trace) my asterisk is not able to answer a call which need Prack. My asterisk answer : ' SIP/2.0 420 Bad extension ... Unsupported: 100rel ' Any idea ? It is because I use port forwarding ? Should i have to open other port ? Thanks -- Jean-Baptiste Bellet Ingénieur Développement Lucyde SAS Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex +33 (0)5 34 31 86 36 http://www.lucyde.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call from h323 to SIP
probably you haven't g729 installed in asterisk, use g711 instead, put this in h323.conf and in callmanager place asterisdk gateway in region that will use g711... disallow=all allow=alaw alternatively you can find g729 codecs binaries here: http://kvin.lv/pub/Linux/Asterisk/ nik600 wrote: Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten = 3298,2,Dial(SIP/[EMAIL PROTECTED]) If a make a call to callamanager CISCO that forward to 3298 i read in asterisk console: Log: Verbosity is at least 20 -- Executing Answer(H323/ip$172.z.z.z:4836/14, ) in new stack -- Executing Dial(H323/ip$172.z.z.z:4836/14, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/[EMAIL PROTECTED] is ringing Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to ulaw Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec ... translation path from g729 to slin Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to ulaw Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to slin Dec 15 14:45:13 WARNING[19794]: translate.c:116 ast_translator_build_path: No translator path from alaw to unknown Dec 15 14:45:13 WARNING[19794]: channel.c:1202 queue_frame_to_spies: Cannot build a path from g729 to slin Dec 15 14:45:13 WARNING[19794]: chan_h323.c:614 oh323_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 4/64) Dec 15 14:45:13 WARNING[19794]: chan_sip.c:2572 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Dec 15 14:45:13 WARNING[19794]: translate.c:116 ast_translator_build_path: No translator path from alaw to unknown Dec 15 14:45:13 WARNING[19794]: channel.c:2752 ast_channel_make_compatible: No path to translate from H323/ip$172.z.z.z:4836/14(256) to SIP/193.x.x.x-40455d68(8) Dec 15 14:45:13 WARNING[19794]: app_dial.c:1602 dial_exec_full: Had to drop call because I couldn't make H323/ip$172.z.z.z:4836/14 compatible with SIP/193.x.x.x-40455d68 == Spawn extension (default, 3298, 2) exited non-zero on 'H323/ip$172.z.z.z:4836/14' Why? where am i wrong? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL: CID match and pattern in switch statement
Hi all,
I am using Asterisk 1.2.10 on Debian Sarge and currently I am rewriting my
extensions.conf with ael.
The replacement of the following part makes me mad:
[set-language]
exten = _X./_0031.,1,Set(incoming_call=1|lang=nl)
exten = _X./_0031.,2,Goto(incoming,${EXTEN},1)
exten = _X./_0049.,1,Set(incoming_call=1|lang=de)
exten = _X./_0049.,2,Goto(incoming,${EXTEN},1)
exten = _X.,1,Set(incoming_call=1|lang=en)
exten = _X.,2,Goto(incoming,${EXTEN},1)
First I tried it this way:
context set-language {
_X./_0031. = {
Set(incoming_call=1|lang=nl);
jump [EMAIL PROTECTED];
};
_X./_0049. = {
Set(incoming_call=1|lang=de);
jump [EMAIL PROTECTED];
};
_X. = {
Set(incoming_call=1|lang=en);
jump [EMAIL PROTECTED];
};
};
The CID match did not seem to work so I tried to solve it with a switch
statement:
context set-language {
_X. = {
Set(original_extension=${EXTEN});
switch (${CALLERID(num)}) {
pattern 0031.:
Set(incoming_call=1|lang=nl);
jump [EMAIL PROTECTED];
pattern 0049.:
Set(incoming_call=1|lang=de);
jump [EMAIL PROTECTED];
default:
Set(incoming_call=1|lang=en);
jump [EMAIL PROTECTED];
};
};
};
Unfortunately pattern does not work at all. I would be very happy if someone
can enlighten me what I am doing wrong here.
Are there any other solutions for replacing that part of my extensions.conf?
Is there any AEL documentation available in addition to the information in
the wiki and doc/README.ael?
Regards, Jens
___
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Re: [asterisk-users] Cisco Call Manager 4.0 to Asterisk, Anyone have SIP Reinvite working?
I think, callmanager needs media termination point (mtp) for sip trunk, so rtp stream will always go through callmanager... JR Richardson wrote: Hi All, I haven't started sip traces or debug yet, but was wondering what the deal is with the CCM and reinvite, why it doesn't work with Asterisk (using 1.2.9.1). I can make calls back and forth all day with canreinvite=no, when I try to reinvite an inbound sip call from the CCM with Asterisk server 1 to Asterisk Server 2, I get one-way audio issues. All the RTP ports are configured the same. I remember Cisco Phones and ATA's have some reinvite issues, wondering if the same applies to the CCM and how it handles reinvites? In the wiki example of integrating CCM with Asterisk, the SIP context shows canreinvite=yes, so should this be working ok, maybe I'm doing something wrong? Thanks JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecting outbound trunks
use contex On 12/15/06, Nigel Kendrick [EMAIL PROTECTED] wrote: Hi Folks, Can you point me towards some info on how to specify that certain extensions use specific outbound trunks - we can only set outbound caller ID against the SIP accounts managed by our service provider (we cannot pass CID info to them at the moment although they are promising this facility) and so I'd like to do the following: Extn A + Extn B + Extn C - Outbound via SIP account 1 only Extn D - Outbound via SIP account 2 only I presume there's some grouping arrangement that will do this? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] anyone using metermaid / parked call BLF?
Hi all, I'm using 1.2.9.1, with the metermaid patches to show parking spot status on Snom BLF lights. I see from http://www.asterisk.org/node/97 that the metermaid code has changed substantially since 1.2.9.1. Is anyone successfully using the new metermaid functionality in 1.4.x? I'd like to hear any good/bad experiences before I attempt even a test upgrade... - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call from h323 to SIP
On 12/15/06, Pavel Jezek [EMAIL PROTECTED] wrote: probably you haven't g729 installed in asterisk, use g711 instead, put this in h323.conf and in callmanager place asterisdk gateway in region that will use g711... disallow=all allow=alaw alternatively you can find g729 codecs binaries here: http://kvin.lv/pub/Linux/Asterisk/ I am experiencig the same problem: h323.conf: disallow=all allow=all ; turns on all installed codecs sip.conf: disallow=all; First disallow all codecs ;allow=all ; Allow codecs in order of preference allow=g711 allow=ulaw extension.conf: exten = 3298,1,Set(SIP_CODEC=alaw) exten = 3298,2,Answer exten = 3298,3,Dial(SIP/[EMAIL PROTECTED]) -- Executing Set(H323/ip$172.z.z.z:1630/20, SIP_CODEC=alaw) in new stack -- Executing Answer(H323/ip$172.z.z.z:1630/20, ) in new stack -- Executing Dial(H323/ip$172.1z.z.z:1630/20, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/193.x.x.x-40451408 is ringing -- Got SIP response 606 Not Acceptable back from 193.x.x.x == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'H323/ip$172.z.z.z:1630/20' status is 'NOANSWER' What means 606 ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call from h323 to SIP
in h323.conf you have still: disallow=all allow=all try change to: disallow=all allow=alaw nik600 wrote: On 12/15/06, Pavel Jezek [EMAIL PROTECTED] wrote: probably you haven't g729 installed in asterisk, use g711 instead, put this in h323.conf and in callmanager place asterisdk gateway in region that will use g711... disallow=all allow=alaw alternatively you can find g729 codecs binaries here: http://kvin.lv/pub/Linux/Asterisk/ I am experiencig the same problem: h323.conf: disallow=all allow=all ; turns on all installed codecs sip.conf: disallow=all; First disallow all codecs ;allow=all ; Allow codecs in order of preference allow=g711 allow=ulaw extension.conf: exten = 3298,1,Set(SIP_CODEC=alaw) exten = 3298,2,Answer exten = 3298,3,Dial(SIP/[EMAIL PROTECTED]) -- Executing Set(H323/ip$172.z.z.z:1630/20, SIP_CODEC=alaw) in new stack -- Executing Answer(H323/ip$172.z.z.z:1630/20, ) in new stack -- Executing Dial(H323/ip$172.1z.z.z:1630/20, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/193.x.x.x-40451408 is ringing -- Got SIP response 606 Not Acceptable back from 193.x.x.x == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'H323/ip$172.z.z.z:1630/20' status is 'NOANSWER' What means 606 ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] enum
On Fri, 2006-12-15 at 11:09 -0800, Khaled wrote: Dear Please how can I make a local dns naptr on my system ,ro resolve local calls using enum http://www.oreilly.com/catalog/dns4/ Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call from h323 to SIP
On 12/15/06, Pavel Jezek [EMAIL PROTECTED] wrote: in h323.conf you have still: disallow=all allow=all try change to: disallow=all allow=alaw i've tried but it gives me the same error... -- Got SIP response 606 Not Acceptable back from 193.x.x.x ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware TDM Switching
[EMAIL PROTECTED] wrote: Is this only possible in a hard configuration way? or is it possible that asterisk handles the call and tell the zap channel, now you have to connect this 2 zap channels cross connect to each other? You can tell Asterisk to connect any channel to any channel. This is what Asterisk does when a call comes in and is sent to another port. This is not done in hardware it is done in Asterisk. Hardware cross connect is called DACS and is not done by Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware TDM Switching
Jonathan k. Creasy wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, December 14, 2006 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hardware TDM Switching [EMAIL PROTECTED] wrote: Do anybody know, if there is a way to connect 2 zap-channels with Hardware TDM Switching? It's called DACS. See the /etc/zapata.conf config file sample. He means /etc/zaptel.conf I thinkright? Correct. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware TDM Switching
On Friday 15 December 2006 10:21 am, Eric ManxPower Wieling wrote: Hardware cross connect is called DACS and is not done by Asterisk. Asterisk does support DACS with Zaptel TDM boards. I know that this is done on-card with the multiport TExxx boards, but I'm not sure if the TDM4xx/24xx boards do it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to know who hangup ?
Gregory Duchatelet a écrit : Hi, Using AMI or dial plan, how can i know which leg (channel ?) of a bridged call, hangup ? AMI send 2 hangup events, which have both cause 16 (normal clearing), and the first hangup event is the called leg hangup event, not the one who hangup… Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users adding g in your dial application and the call will go on the extension when the callee hangup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware TDM Switching
Andrew Kohlsmith wrote: On Friday 15 December 2006 10:21 am, Eric ManxPower Wieling wrote: Hardware cross connect is called DACS and is not done by Asterisk. Asterisk does support DACS with Zaptel TDM boards. I know that this is done on-card with the multiport TExxx boards, but I'm not sure if the TDM4xx/24xx boards do it. DACS is not done in Asterisk. DACS is done in the Zaptel drivers. In fact, you can do DACS without Asterisk even being installed on the system. Also the channels that are DACS'd are not even accessible to Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware TDM Switching
On Friday 15 December 2006 10:52 am, Eric ManxPower Wieling wrote: DACS is not done in Asterisk. DACS is done in the Zaptel drivers. In fact, you can do DACS without Asterisk even being installed on the system. Also the channels that are DACS'd are not even accessible to Asterisk. You're absolutely right, I keep the two in the same basket in my head. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware TDM Switching (Out Of Office - on vacation)
I will be out the office on vacation. asterisk-users@lists.digium.com 12/15/06 11:25 On Friday 15 December 2006 10:52 am, Eric ManxPower Wieling wrote: DACS is not done in Asterisk. DACS is done in the Zaptel drivers. In fact, you can do DACS without Asterisk even being installed on the system. Also the channels that are DACS'd are not even accessible to Asterisk. You're absolutely right, I keep the two in the same basket in my head. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to know who hangup ?
adding g in your dial application and the call will go on the extension
when the callee hangup
Yes, i could also use h extension, but how to know which one hangup first
? ${HANGUPCAUSE} always say 16 (Normal clearing), and ${CHANNEL} is set to
the current channel in the dial plan...
Greg
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Re: [asterisk-users] call from h323 to SIP
1) why you Answer() before Dial() 2) try Dial(SIP/user) instead of Dial(SIP/[EMAIL PROTECTED]) asterisk knows, what IP has peer (sip show peers) 3) try call echo() test aplication from callmanager phone nik600 wrote: On 12/15/06, Pavel Jezek [EMAIL PROTECTED] wrote: in h323.conf you have still: disallow=all allow=all try change to: disallow=all allow=alaw i've tried but it gives me the same error... -- Got SIP response 606 Not Acceptable back from 193.x.x.x ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco Call Manager 4.0 to Asterisk, Anyone haveSIP Reinvite working?
Pavel wrote: I think, callmanager needs media termination point (mtp) for sip trunk, so rtp stream will always go through callmanager... That is true for CCM 4.X, so SIP works with CCM 4.X, but is far from ideal. As of CCM 5.X added RFC 2833 support to the SCCP endpoints, so a MTP is not required and your not stuck with just ULAW for the codec Now whether improved SIP support is enough to justify the big jump to 5.x (Windows to Linux), is another issue... Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP DTMF not acted on for features in 1.4.0b3
Asterisk seems to be ignoring DTMF for features in Asterisk 1.4.0b3 My SNOM sends the dtmf-relay and Asterisk gets it because I can see it in the sip debug. However, once seen, Asterisk doesn't actually do anything about it. I've checked features and that seems fine. Is this a bug or something that I've screwed up? For the record, here's the features setting: asterisk*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer *2 One Touch Monitor *1 Disconnect Call * * Park Call #72 Dynamic Feature Default Current --- --- --- testfeature no def #9 Call parking Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-720 asterisk*CLI and here's a SIP trace of me pressing '*' during a call (which according to my features should Disconnect the Call. asterisk*CLI --- SIP read from 192.168.1.12:5060 --- INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK-llnm8m4u2wef;rport From: Russell 112 sip:[EMAIL PROTECTED];tag=tmyszljbna To: sip:[EMAIL PROTECTED];tag=as0b7389e4 Call-ID: [EMAIL PROTECTED] CSeq: 14 INFO Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;line=gv8x1x75;flow-id=1 User-Agent: snom360/6.5.1 Content-Type: application/dtmf-relay Content-Length: 22 Signal=* Duration=160 - --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * asterisk*CLI --- Transmitting (no NAT) to 192.168.1.12:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK-llnm8m4u2wef;received=192.168.1.12;rport=5060 From: Russell 112 sip:[EMAIL PROTECTED];tag=tmyszljbna To: sip:[EMAIL PROTECTED];tag=as0b7389e4 Call-ID: [EMAIL PROTECTED] CSeq: 14 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/112-0070a2c0] asterisk*CLI Can anyone suggest what's wrong here? Thanks. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying to forward calls by using the Callee's context as the forward dial context
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally be allowed to dial within their own context. (I
don't want a local call only person forwarding to a long dist number,
for example.) I'm able to get the channel context for SIP devices but
not for IAX or Zap Devices. I need some pointers on getting IAXPEER to
work and how to handle getting the ZAP context info. If there's an
easier way, I'm all ears. Thanks.
; #Set Some Variables
exten = s,1,Set(DEVICE=${ARG1}) ;i.e. SIP/johns_phone
exten = s,n,Set(Protocol=${CUT(DEVICE,/,1)}) ;Parse out SIP
exten = s,n,Set(Phone=${CUT(DEVICE,/,2)}) ;Parse out johns_phone
;Stuff omitted for some amout of brevity
; #Make Forward Calls##
; We only want people to be able to forward to numbers they could
normally call
; We'll have to somehow get their dialing contexts from the channel conf
files.
exten = s,n(Forward),NoOp()
exten = s,n,GotoIf($[${Protocol} = SIP]?SIPDev)
exten = s,n,GotoIf($[${Protocol} = IAX2]?IAXDev)
exten = s,n,Goto(ZapDev)
;ok, they are an IAX device so use IAXPEER
exten =
s,n(IAXDev),Set(CalledUsersContext=${IAXPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an SIP device so use SIPPEER
exten =
s,n(SIPDev),Set(CalledUsersContext=${SIPPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an Zap device so use... Uh.
exten = s,n(ZapDev),NoOp( I have no clue how to get the zap channel's
context...)
exten = s,n(dial_time),NoOp(== Chan Type
${Protocol})
exten = s,n,NoOp(== Chan Name ${Phone})
exten = s,n,NoOp(== Channel User's context
${CalledUsersContext})
exten = s,n,Dial(Local/[EMAIL PROTECTED]/n)
Results at console on verbosity 9:
SIPPEER() Works as advertised when I dial a SIP phone which has been
call forwarded
-- Executing NoOp(Zap/1-1, == Chan Type
SIP) in new stack
-- Executing NoOp(Zap/1-1, == Chan Name
jf_linksys) in new stack
-- Executing NoOp(Zap/1-1, == Channel Users
context longdistance_users) in new stack
-- Executing Dial(Zap/1-1, Local/[EMAIL PROTECTED]/n)
in new stack
IAXPEER() Seems to be broken or I don't know how to use it properly.
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Type IAX2) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Name johns_pc) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Channel Users context ) in new stack
-- Executing Dial(SIP/jf_linksys-08f20548, Local/5551212@/n) in
new stack
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[asterisk-users] Sip port= not working
I am using a month old svn version of asterisk 1.2 . I have set bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show peer it shows port 5091 for peer but asterisk isnt listening on port 5091 at all . I tried both port=5091 as well as binport=5091 but asterisk does not listen on port 5091 . What am i doing wrong ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to forward calls by using the Callee's context as the forward dial context
Hi John,
I would try to use on sip.conf and iax.conf and zapata.conf:
on every user (friend or whatever) defined add this:
[useraccount]
setvar=mycontext=yourcontext
--
This variable will become available for every user, so you just need to use
it in your dialplan in extensions.conf
Noop(User context:$mycontext)
This is just an idea, please give some feedback if it helped and some how
you will test if the forward number is valid or not :)
Probably isn't hard , but is not clear yet for me and i'm busy :)
Best Regards,
Marco Mouta
On 12/15/06, John French [EMAIL PROTECTED] wrote:
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally be allowed to dial within their own context. (I
don't want a local call only person forwarding to a long dist number,
for example.) I'm able to get the channel context for SIP devices but
not for IAX or Zap Devices. I need some pointers on getting IAXPEER to
work and how to handle getting the ZAP context info. If there's an
easier way, I'm all ears. Thanks.
; #Set Some Variables
exten = s,1,Set(DEVICE=${ARG1}) ;i.e. SIP/johns_phone
exten = s,n,Set(Protocol=${CUT(DEVICE,/,1)}) ;Parse out SIP
exten = s,n,Set(Phone=${CUT(DEVICE,/,2)}) ;Parse out johns_phone
;Stuff omitted for some amout of brevity
; #Make Forward Calls##
; We only want people to be able to forward to numbers they could
normally call
; We'll have to somehow get their dialing contexts from the channel conf
files.
exten = s,n(Forward),NoOp()
exten = s,n,GotoIf($[${Protocol} = SIP]?SIPDev)
exten = s,n,GotoIf($[${Protocol} = IAX2]?IAXDev)
exten = s,n,Goto(ZapDev)
;ok, they are an IAX device so use IAXPEER
exten =
s,n(IAXDev),Set(CalledUsersContext=${IAXPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an SIP device so use SIPPEER
exten =
s,n(SIPDev),Set(CalledUsersContext=${SIPPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an Zap device so use... Uh.
exten = s,n(ZapDev),NoOp( I have no clue how to get the zap channel's
context...)
exten = s,n(dial_time),NoOp(== Chan Type
${Protocol})
exten = s,n,NoOp(== Chan Name ${Phone})
exten = s,n,NoOp(== Channel User's context
${CalledUsersContext})
exten = s,n,Dial(Local/[EMAIL PROTECTED]/n)
Results at console on verbosity 9:
SIPPEER() Works as advertised when I dial a SIP phone which has been
call forwarded
-- Executing NoOp(Zap/1-1, == Chan Type
SIP) in new stack
-- Executing NoOp(Zap/1-1, == Chan Name
jf_linksys) in new stack
-- Executing NoOp(Zap/1-1, == Channel Users
context longdistance_users) in new stack
-- Executing Dial(Zap/1-1, Local/[EMAIL PROTECTED]/n)
in new stack
IAXPEER() Seems to be broken or I don't know how to use it properly.
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Type IAX2) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Name johns_pc) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Channel Users context ) in new stack
-- Executing Dial(SIP/jf_linksys-08f20548, Local/5551212@/n) in
new stack
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Re: [asterisk-users] fxotune unable to set impedence
Yuan LIU wrote: I just didn't want to accept fxotune.c's claim about working only with TDM. Several other users indicated that they were not able to tune X100P. There's also a README.debian note that specifically indicated exclusion of X100P. fxotune is written to change register values on a specific Silicon Labs chip, which is used in the TDM400 FXO modules. No X100P uses this chip, (and the chips they use do not have the feature used), so fxotune does nothing. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zapata.conf channel variable question
The setvar command below works fine in iax.conf and in sip.conf but not here in zaptel.conf. I need it to retrieve info from the AstDB. Advice is apreciated, can't seem to find an answer. ; define channels group=1 context=longdistance_users signalling=fxo_ks ;FXO Sig for Phone callerid=John French 103 mailbox=101 callwaiting=yes threewaycalling=yes transfer=yes channel = 1 setvar=USER=analogPhone ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to forward calls by using the Callee's context as the forward dial context
Hi John, I´m very interested into this call forwarding capabilities and
I solved this problem filtering on the web-script (in my case, php) the
number the user can intert on the database. (I know it´s not an asterisk
solution).
There is an issue that I couldn´t handle. When I forward the call, I
want to charge the user that the call was made FOR. How are you dealing
with that? Going direct to the point, I just need to know - a tip would
be apreciated either - how to translate/replace the FROM field of the
forwarded call.
Rgds, Ricardo.
John French wrote:
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally be allowed to dial within their own context. (I
don't want a local call only person forwarding to a long dist number,
for example.) I'm able to get the channel context for SIP devices but
not for IAX or Zap Devices. I need some pointers on getting IAXPEER to
work and how to handle getting the ZAP context info. If there's an
easier way, I'm all ears. Thanks.
; #Set Some Variables
exten = s,1,Set(DEVICE=${ARG1}) ;i.e. SIP/johns_phone
exten = s,n,Set(Protocol=${CUT(DEVICE,/,1)}) ;Parse out SIP
exten = s,n,Set(Phone=${CUT(DEVICE,/,2)}) ;Parse out johns_phone
;Stuff omitted for some amout of brevity
; #Make Forward Calls##
; We only want people to be able to forward to numbers they could
normally call
; We'll have to somehow get their dialing contexts from the channel conf
files.
exten = s,n(Forward),NoOp()
exten = s,n,GotoIf($[${Protocol} = SIP]?SIPDev)
exten = s,n,GotoIf($[${Protocol} = IAX2]?IAXDev)
exten = s,n,Goto(ZapDev)
;ok, they are an IAX device so use IAXPEER
exten =
s,n(IAXDev),Set(CalledUsersContext=${IAXPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an SIP device so use SIPPEER
exten =
s,n(SIPDev),Set(CalledUsersContext=${SIPPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an Zap device so use... Uh.
exten = s,n(ZapDev),NoOp( I have no clue how to get the zap channel's
context...)
exten = s,n(dial_time),NoOp(== Chan Type
${Protocol})
exten = s,n,NoOp(== Chan Name ${Phone})
exten = s,n,NoOp(== Channel User's context
${CalledUsersContext})
exten = s,n,Dial(Local/[EMAIL PROTECTED]/n)
Results at console on verbosity 9:
SIPPEER() Works as advertised when I dial a SIP phone which has been
call forwarded
-- Executing NoOp(Zap/1-1, == Chan Type
SIP) in new stack
-- Executing NoOp(Zap/1-1, == Chan Name
jf_linksys) in new stack
-- Executing NoOp(Zap/1-1, == Channel Users
context longdistance_users) in new stack
-- Executing Dial(Zap/1-1, Local/[EMAIL PROTECTED]/n)
in new stack
IAXPEER() Seems to be broken or I don't know how to use it properly.
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Type IAX2) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Name johns_pc) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Channel Users context ) in new stack
-- Executing Dial(SIP/jf_linksys-08f20548, Local/5551212@/n) in
new stack
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Re: [asterisk-users] Show agent queue status on the phone?
some idea, how to make BLF working on ci$co 7961 (sip)? Steve Langstaff wrote: I've not used the Cisco kit for this, but you might try adding 'hints' to your agent extensions, and then defining a BLF button to subscribe to this. e.g. If you have an agent with ID 1001, add this to extensions.conf (or equivalent)... exten = 1001,hint,Agent/1001 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Trumbull Sent: 14 December 2006 18:04 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Show agent queue status on the phone? Hi All, Is it possible to show an agent's queue status on the phone? For example, in our current non-asterisk PBX, if a member of a call queue does not answer the phone when a queue call is sent to them, they go to a 'not ready' status, and this is indicated on their phone. So when they return to their desk, they can see that they are not ready, so they hit a button to put themselves back into a ready status. I can accomplish the 'not ready' functionality by using the PauseQueueMember function, but now I need to somehow display the pause/unpause status on the phone so the staff member knows if they got paused. Does anybody know how to do this? This might be phone specific, so I'll mention that we are using Cisco 7961G's. Thanks -Kevin Trumbull ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call from h323 to SIP
On 12/15/06, Pavel Jezek [EMAIL PROTECTED] wrote: 1) why you Answer() before Dial() sorry, it is a my error 2) try Dial(SIP/user) instead of Dial(SIP/[EMAIL PROTECTED]) asterisk knows, what IP has peer (sip show peers) no, because the user isn't registered on asterisk server. asterisk is at 193.y.y.y and the sip user at 193.x.x.x ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call from h323 to SIP
nik600 wrote: Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. The incoming call is in the g.729 format, you should be able to fix this in cisco call manager. If not, make sure that the SIP target can accept a g.729 call. Failing that buy a license for the codec. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten = 3298,2,Dial(SIP/[EMAIL PROTECTED]) If a make a call to callamanager CISCO that forward to 3298 i read in asterisk console: Log: Verbosity is at least 20 -- Executing Answer(H323/ip$172.z.z.z:4836/14, ) in new stack -- Executing Dial(H323/ip$172.z.z.z:4836/14, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/[EMAIL PROTECTED] is ringing Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to ulaw Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec ... translation path from g729 to slin Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to ulaw Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to slin Dec 15 14:45:13 WARNING[19794]: translate.c:116 ast_translator_build_path: No translator path from alaw to unknown Dec 15 14:45:13 WARNING[19794]: channel.c:1202 queue_frame_to_spies: Cannot build a path from g729 to slin Dec 15 14:45:13 WARNING[19794]: chan_h323.c:614 oh323_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 4/64) Dec 15 14:45:13 WARNING[19794]: chan_sip.c:2572 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Dec 15 14:45:13 WARNING[19794]: translate.c:116 ast_translator_build_path: No translator path from alaw to unknown Dec 15 14:45:13 WARNING[19794]: channel.c:2752 ast_channel_make_compatible: No path to translate from H323/ip$172.z.z.z:4836/14(256) to SIP/193.x.x.x-40455d68(8) Dec 15 14:45:13 WARNING[19794]: app_dial.c:1602 dial_exec_full: Had to drop call because I couldn't make H323/ip$172.z.z.z:4836/14 compatible with SIP/193.x.x.x-40455d68 == Spawn extension (default, 3298, 2) exited non-zero on 'H323/ip$172.z.z.z:4836/14' Why? where am i wrong? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What's up with DATETIME and TIMESTAMP in Asterisk 1.4beta3 ?
Hello,
In Asterisk 1.4 beta 3, the UPGRADE.txt file says:
Variables:
* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP},
${ACCOUNTCODE},
and ${LANGUAGE} have all been deprecated in favor of their related
dialplan
functions. You are encouraged to move towards the associated dialplan
function, as these variables will be removed in a future release.
However, neither the function or application for either of TIMESTAMP or
DATETIME seems to work in 1.4beta3...
exten = *333,1,NoOp(DATETIME() : ${DATETIME()})
exten = *333,n,NoOp(DATETIME : ${DATETIME})
exten = *333,n,NoOp(TIMESTAMP() : ${TIMESTAMP()})
exten = *333,n,NoOp(TIMESTAMP : ${TIMESTAMP})
Asterisk 1.2.9.1:
-
Dec 15 12:56:26 ERROR[26373]: pbx.c:1383 ast_func_read: Function
DATETIME not registered
-- Executing NoOp(channel, DATETIME() : 0) in new stack
-- Executing NoOp(channel, DATETIME : 20061215-12:56:26) in
new stack
Dec 15 12:56:26 ERROR[26373]: pbx.c:1383 ast_func_read: Function
TIMESTAMP not registered
-- Executing NoOp(channel, TIMESTAMP() : 0) in new stack
-- Executing NoOp(channel, TIMESTAMP : 20061215-125626) in new
stack
Asterisk 1.4.0-beta3:
-
[Dec 15 13:59:52] ERROR[28236]: pbx.c:1497 ast_func_read: Function
DATETIME not registered
-- Executing [*333@context:1] NoOp(channel, DATETIME() : )
in new stack
-- Executing [*333@context:2] NoOp(channel, DATETIME : ) in
new stack
[Dec 15 13:59:52] ERROR[28236]: pbx.c:1497 ast_func_read: Function
TIMESTAMP not registered
-- Executing [*333@context:3] NoOp(channel, TIMESTAMP() : )
in new stack
-- Executing [*333@context:4] NoOp(channel, TIMESTAMP : ) in
new stack
Any ideas?
Thanks,
Alvin
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[asterisk-users] Fast Busy Followup
So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone for your help with this matter, Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iptables rule help
Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under extensions is not working .. asterisk doesnt listen to port 5091 .. but if i put in general section of sip.confthen it works but then asterisk wont listen on 5060 . How can i use iptables in this situation ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iptables rule help
Mail list wrote: Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under extensions is not working .. asterisk doesnt listen to port 5091 .. but if i put in general section of sip.confthen it works but then asterisk wont listen on 5060 . How can i use iptables in this situation ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users RTP ports blocked? (rtp.conf) signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's up with DATETIME and TIMESTAMP in Asterisk 1.4beta3 ?
Am Freitag, den 15.12.2006, 13:08 -0600 schrieb Alvin Austin:
Hello,
In Asterisk 1.4 beta 3, the UPGRADE.txt file says:
Variables:
* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP},
${ACCOUNTCODE},
and ${LANGUAGE} have all been deprecated in favor of their related
dialplan
functions. You are encouraged to move towards the associated dialplan
function, as these variables will be removed in a future release.
However, neither the function or application for either of TIMESTAMP or
DATETIME seems to work in 1.4beta3...
exten = *333,1,NoOp(DATETIME() : ${DATETIME()})
exten = *333,n,NoOp(DATETIME : ${DATETIME})
exten = *333,n,NoOp(TIMESTAMP() : ${TIMESTAMP()})
exten = *333,n,NoOp(TIMESTAMP : ${TIMESTAMP})
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+strftime
will probably contain the information you want. To sum up, DATETIME and
TIMESTAMP are gone, use the ${EPOCH} for the seconds since
1970-01-01,00:00:00 and the STRFTIME to format that data.
STRPTIME can be used to calculate the epoch value of any
date-time-string.
HTH
Anselm
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[asterisk-users] DTMF Tone Issues
I have 1.2.12.1 Voicepulse using IAX I get about 30-40% issues with not having the DTMF tones work. I have 3 questions #1. Voicepulse says they are sending them, Is there some setting I can adjust to make sure my end is working? #2. I have set the Dialplan to play a sound Operator then go to a SIP phone. I would like it to write Caller ID Time to a file I can read and find out exactly how many people are getting to that point. #3. If it is Voicepulses fault. Who else might you suggest for my numbers to be ported to and handle my phone calls Thanks Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip port= not working
port= specifies the REMOTE port. You can't have multiple bindport= and it must be in [general] Mail list wrote: I am using a month old svn version of asterisk 1.2 . I have set bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show peer it shows port 5091 for peer but asterisk isnt listening on port 5091 at all . I tried both port=5091 as well as binport=5091 but asterisk does not listen on port 5091 . What am i doing wrong ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iptables rule help
Are you in the US? If so, such blocking is not legal, and you should file a complaint with the FCC John Novack Mail list wrote: Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 http://127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under extensions is not working .. asterisk doesnt listen to port 5091 .. but if i put in general section of sip.conf then it works but then asterisk wont listen on 5060 . How can i use iptables in this situation ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zapata.conf channel variable question
On Fri, Dec 15, 2006 at 12:10:40PM -0600, John French wrote: The setvar command It is not a dialplan command. It is a configuration key. below works fine in iax.conf and in sip.conf but not here in zaptel.conf. I need it to retrieve info from the AstDB. Advice is apreciated, can't seem to find an answer. Two problems: 1. setvar does not seem to be supported with zaptel channels. and: ; define channels group=1 context=longdistance_users signalling=fxo_ks ;FXO Sig for Phone callerid=John French 103 mailbox=101 callwaiting=yes threewaycalling=yes transfer=yes channel = 1 setvar=USER=analogPhone If you wanted it to take effect you needed to put it before the 'channel' line. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to forward calls by using the Callee's context as the forward dial context
forking CDR could help Ricardo.
On 12/15/06, Ricardo Martins [EMAIL PROTECTED] wrote:
Hi John, I´m very interested into this call forwarding capabilities and
I solved this problem filtering on the web-script (in my case, php) the
number the user can intert on the database. (I know it´s not an asterisk
solution).
There is an issue that I couldn´t handle. When I forward the call, I
want to charge the user that the call was made FOR. How are you dealing
with that? Going direct to the point, I just need to know - a tip would
be apreciated either - how to translate/replace the FROM field of the
forwarded call.
Rgds, Ricardo.
John French wrote:
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally be allowed to dial within their own context. (I
don't want a local call only person forwarding to a long dist number,
for example.) I'm able to get the channel context for SIP devices but
not for IAX or Zap Devices. I need some pointers on getting IAXPEER to
work and how to handle getting the ZAP context info. If there's an
easier way, I'm all ears. Thanks.
; #Set Some Variables
exten = s,1,Set(DEVICE=${ARG1}) ;i.e. SIP/johns_phone
exten = s,n,Set(Protocol=${CUT(DEVICE,/,1)}) ;Parse out SIP
exten = s,n,Set(Phone=${CUT(DEVICE,/,2)}) ;Parse out johns_phone
;Stuff omitted for some amout of brevity
; #Make Forward Calls##
; We only want people to be able to forward to numbers they could
normally call
; We'll have to somehow get their dialing contexts from the channel conf
files.
exten = s,n(Forward),NoOp()
exten = s,n,GotoIf($[${Protocol} = SIP]?SIPDev)
exten = s,n,GotoIf($[${Protocol} = IAX2]?IAXDev)
exten = s,n,Goto(ZapDev)
;ok, they are an IAX device so use IAXPEER
exten =
s,n(IAXDev),Set(CalledUsersContext=${IAXPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an SIP device so use SIPPEER
exten =
s,n(SIPDev),Set(CalledUsersContext=${SIPPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an Zap device so use... Uh.
exten = s,n(ZapDev),NoOp( I have no clue how to get the zap channel's
context...)
exten = s,n(dial_time),NoOp(== Chan Type
${Protocol})
exten = s,n,NoOp(== Chan Name ${Phone})
exten = s,n,NoOp(== Channel User's context
${CalledUsersContext})
exten = s,n,Dial(Local/[EMAIL PROTECTED]/n)
Results at console on verbosity 9:
SIPPEER() Works as advertised when I dial a SIP phone which has been
call forwarded
-- Executing NoOp(Zap/1-1, == Chan Type
SIP) in new stack
-- Executing NoOp(Zap/1-1, == Chan Name
jf_linksys) in new stack
-- Executing NoOp(Zap/1-1, == Channel Users
context longdistance_users) in new stack
-- Executing Dial(Zap/1-1, Local/[EMAIL PROTECTED]/n)
in new stack
IAXPEER() Seems to be broken or I don't know how to use it properly.
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Type IAX2) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Name johns_pc) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Channel Users context ) in new stack
-- Executing Dial(SIP/jf_linksys-08f20548, Local/5551212@/n) in
new stack
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Re: [asterisk-users] Iptables rule help
well that should map incoming packets to 5091 to 5060, but may not rewrite [new] outbound packets from 5060 to 5091, which your isp may be blocking. an iptables SNAT or MASQUERADE might help you there. i'm not positive on if this would be needed or not. more importantly, however, if your isp is blocking all outgoing traffic to 5060, it won't get to your softphone anyway, unless you also configure that end to also not use 5060. and if you're reconfiguring ports on the softphone end anyway, why not just put 5091 in there, 5091 in sip.conf's bindport, and not mess with iptables at all? another option might be that your isp is blocking rtp as well. can you see what the asterisk console is doing when you attempt such calls? and/or tcpdump? -tcl. On Sat, 16 Dec 2006, Mail list wrote: Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under extensions is not working .. asterisk doesnt listen to port 5091 .. but if i put in general section of sip.confthen it works but then asterisk wont listen on 5060 . How can i use iptables in this situation ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iptables rule help
I am sure rtp ports arent blocked .. On 16/12/06, Derek Whitten [EMAIL PROTECTED] wrote: Mail list wrote: Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under extensions is not working .. asterisk doesnt listen to port 5091 .. but if i put in general section of sip.confthen it works but then asterisk wont listen on 5060 . How can i use iptables in this situation ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users RTP ports blocked? (rtp.conf) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Fast Busy Followup
What kinf of line do your DIDs come in on? How many spans do you have configured and where do they go? Telco/legacy PBX? Does span 2 have a context defined? -- -- Steven http://www.glimasoutheast.org Rob Schall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone for your help with this matter, Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip port= not working
Yes i read that on voip-info wiki but i have bindport = under device (extension) which should make that extension work on other port but its not working . :( On 16/12/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: port= specifies the REMOTE port. You can't have multiple bindport= and it must be in [general] Mail list wrote: I am using a month old svn version of asterisk 1.2 . I have set bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show peer it shows port 5091 for peer but asterisk isnt listening on port 5091 at all . I tried both port=5091 as well as binport=5091 but asterisk does not listen on port 5091 . What am i doing wrong ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Fast Busy Followup
I've set it up as... span=2,1,0,esf,b8zs bchan=6-27 dchan=28 It is a paetec full pri t1. Does this help with the diagnosis, or do you need more info? Rob Steven wrote: What kinf of line do your DIDs come in on? How many spans do you have configured and where do they go? Telco/legacy PBX? Does span 2 have a context defined? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iptables rule help
I have shifted asterisk port to 5091 . Now i am able to register properly using sjphone but still when dialing number it keep on showing calling .. and do not go ahead . I change asterisk's rtp ports too but still i am unable to make call . My other softphone on different internet isp is working properly . :( On 16/12/06, Tim C. Lewis [EMAIL PROTECTED] wrote: well that should map incoming packets to 5091 to 5060, but may not rewrite [new] outbound packets from 5060 to 5091, which your isp may be blocking. an iptables SNAT or MASQUERADE might help you there. i'm not positive on if this would be needed or not. more importantly, however, if your isp is blocking all outgoing traffic to 5060, it won't get to your softphone anyway, unless you also configure that end to also not use 5060. and if you're reconfiguring ports on the softphone end anyway, why not just put 5091 in there, 5091 in sip.conf's bindport, and not mess with iptables at all? another option might be that your isp is blocking rtp as well. can you see what the asterisk console is doing when you attempt such calls? and/or tcpdump? -tcl. On Sat, 16 Dec 2006, Mail list wrote: Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under extensions is not working .. asterisk doesnt listen to port 5091 .. but if i put in general section of sip.confthen it works but then asterisk wont listen on 5060 . How can i use iptables in this situation ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iptables rule help
Am Samstag, den 16.12.2006, 02:32 +0530 schrieb Vicky: I have shifted asterisk port to 5091 . Now i am able to register properly using sjphone but still when dialing number it keep on showing calling .. and do not go ahead . I change asterisk's rtp ports too but still i am unable to make call . My other softphone on different internet isp is working properly . :( Can your softphone reach a voicemail() extension, or echotest? Will that work with audio in both directions? And then, I am not sure wether I understand your setup correctly. Are you trying Asterisk [portblocked line] ISP. Internet. Softphone ? In that case, local tests like those mentioned above will help to rule out wether problems are on the internet part or possibly on a PRI or whatever connected to your asterisk, over which you would like to dial out. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good Commercial Grade Service Provider?
We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH Between Asterisk Servers
Scenario: A call is sent from one Asterisk system to another with IAX. The remote Asterisk system runs the Queue application, which then starts to play a different music on hold class then the standard 'default'. The console on this system displays: -- Executing Queue(IAX2/xxx.yyy.142.203:4569-4, demo_QMain|t|||60) in new stack -- Started music on hold, class 'demo_MainOffice', on IAX2/xxx.yyy.142.203:4569-4 -- Called SIP/2943367 -- Called SIP/2943368 -- SIP/2943367-1bb8 is ringing -- SIP/2943368-537f is ringing However, on the first Asterisk system, we see this on the console: -- Called dundiapps:[EMAIL PROTECTED]/demo_EMain -- Call accepted by xxx.yyy.142.204 (format g729) -- Format for call is g729 -- Started music on hold, class 'default', on IAX2/xxx.yyy.142.203:4569-5 The music on hold class in use is not being conveyed back to the original Asterisk system. Please don't tell me this is a limitation. That would be very very bad. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialing via SIP URI
Does anyone on-list have experience doing this? I'm curious about setting it up. I own a domain and might like to try making sip:[EMAIL PROTECTED] a workable idea. Is this just an experimental thing, or might it be really usefull...say for video calling? Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iptables rule help
On Fri, 15 Dec 2006, John Novack wrote: Are you in the US? If so, such blocking is not legal I'd like to see a citation for that. ISPs aren't common carriers and aren't required to carry specific types of traffic. and you should file a complaint with the FCC The FCC regulates common carriers. ISP's aren't. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Motherboard 3.3V PCI for TE412P
Hi all Does anyone know of any motherboards with PCI slots that can take the TE412P card? Is there such a MB for Athlon 64 or P4 procs? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iptables rule help
Actually port block is on softphone side and not on asterisk server's internet connection .I put this in iptables of asterisk server iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j REDIRECT --to-port 127.0.0.1:5060 server is listening on port 5060 Now strange part is everything s working .. but asterisk is not detecting hangup . I make call on softphone .. call goes everything works fine but when i hang softphone .. i can see on asterisk that call is still going on .,... and this is not a problem of softphone i am sure of that :( On 16/12/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Samstag, den 16.12.2006, 02:32 +0530 schrieb Vicky: I have shifted asterisk port to 5091 . Now i am able to register properly using sjphone but still when dialing number it keep on showing calling .. and do not go ahead . I change asterisk's rtp ports too but still i am unable to make call . My other softphone on different internet isp is working properly . :( Can your softphone reach a voicemail() extension, or echotest? Will that work with audio in both directions? And then, I am not sure wether I understand your setup correctly. Are you trying Asterisk [portblocked line] ISP. Internet. Softphone ? In that case, local tests like those mentioned above will help to rule out wether problems are on the internet part or possibly on a PRI or whatever connected to your asterisk, over which you would like to dial out. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fast Busy Followup
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Friday, December 15, 2006 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fast Busy Followup So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone for your help with this matter, Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have the same problem with a span from Bell Canada. After time, calls begin to fail with the same Ring requested ... error message. I found that if I restart Zaptel and Asterisk, that the problem goes away for a while. Ron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iptables rule help
Actually port block is on softphone side and not on asterisk server's internet connection .I put this in iptables of asterisk server iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j REDIRECT --to-port 127.0.0.1:5060 server is listening on port 5060 Now strange part is everything s working .. but asterisk is not detecting hangup . I make call on softphone .. call goes everything works fine but when i hang softphone .. i can see on asterisk that call is still going on .,... and this is not a problem of softphone i am sure of that :( On 16/12/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Samstag, den 16.12.2006, 02:32 +0530 schrieb Vicky: I have shifted asterisk port to 5091 . Now i am able to register properly using sjphone but still when dialing number it keep on showing calling .. and do not go ahead . I change asterisk's rtp ports too but still i am unable to make call . My other softphone on different internet isp is working properly . :( Can your softphone reach a voicemail() extension, or echotest? Will that work with audio in both directions? And then, I am not sure wether I understand your setup correctly. Are you trying Asterisk [portblocked line] ISP. Internet. Softphone ? In that case, local tests like those mentioned above will help to rule out wether problems are on the internet part or possibly on a PRI or whatever connected to your asterisk, over which you would like to dial out. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax2 softphone attended transfers
Is there any good iax2 softphone capable of attended transfer ( like sjphone for sip ) . ? I tried iaxcomm and idefisk both seems unable to handle attended transfers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] International Provider
Hello everybody Anyone know a good carrier of voip for international calls? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sipura question
I have a Sipura 3k connected to Asterisk 1.2. The web interface for the Sipura, on the PSTN line tab lists VoIP User 1 Auth ID: asterisk and Dial Plan 8: (S0:66610) How do I put the Dial Plan 8 information in sip.conf or extensions.conf? Is 66610 a sip extension in sip.conf or a context in extensions.conf? Or other? Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Commercial Grade Service Provider?
On 12/15/06, Paul Connolly [EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? Check at teliax.com. I think they allow at least 10, maybe more. voipstreet.com allows at least 20. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura question
Larry Alkoff wrote: I have a Sipura 3k connected to Asterisk 1.2. All I want to do here is have incoming PSTN calls ring POTS phones connected to the Sipura. The web interface for the Sipura, on the PSTN line tab lists VoIP User 1 Auth ID: asterisk and Dial Plan 8: (S0:66610) How do I put the Dial Plan 8 information in sip.conf or extensions.conf? Is 66610 a sip extension in sip.conf or a context in extensions.conf? Or other? Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Boot load wcfxo does not configure self under Ubuntu 6
When booting Ubuntu 6.06.1 (Linux 2.6.15-27-386), wcfxo would load but not configure. I have three ways to manually force wcfxo to configure: 1) ztcfg, 2) modprobe -f wcfxo, or of course 3) unload and reload wcfxo. Each works equally well. As a workaround, now I have to put ztcfg in rc.local, which Debian doesn't favour. I had compiled the same zaptel under Ubuntu 5.02.0 on two different machines with same card and didn't have this problem. Anyone else experiencing the same? As tempting as it is to blame /etc/modprobe.d, I figure that if a unload and reload wcfxo correctly configures the module, the method must be working. Or has this anything to do with the debian_version note testing/unstable? (Ubuntu 6.06.1 is supposed to be a formal release.) Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IBM Server / USB Ports
I see that the digium card doesn't share the IRQ however Digium has recommended diabled USB still... additionally the Digium card is on 169 which isn't a valid IRQ.. how can I find out what it is sharing with? lspci -vb will give you the irq as seen by the cards on the PCI bus -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.21/589 - Release Date: 12/15/2006 5:10 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztmonitor displays full bar when idle
Hardware is an SM56 card (X100P clone). When the line hangs up, ztmonitor displays full bar (or whatever maximum allowed by rxgain) in RX. It only drops zero when the line picks up (and remote was silent). Is this something of concern? The zap channel seems to work despite echo. Additionally, what's the objective of tuning with ztmonitor? I mean, what would indicate an optimal level? (Because display of ztmonitor depends on the volume of the remote speaker - I'm using demo sounds from another Asterisk, how do I tell?) Documents I see only mentioned RX, should I be concerned about TX? What's a good indicator of an optimal txgain? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bandwidth requirements for 1, 000, 000 minutes a month
This may expose my ignorance, but here goes :) I've been asked to figure out how much bandwidth would be needed to handle 1,000,000 minutes a month. Here's the environment: ) All calls are received via SIP. ) All calls use the ulaw codec. ) Calls average 10 minutes in duration. ) The busiest hour will account for 10% of the daily total. This is how I'm figuring it... Casual observation shows that SIP setup and teardown takes about 26 UDP packets. Assuming the packets are full (512 bytes) this adds up to about 13 kilo-bytes for each call. I've heard that ulaw (including overhead) is supposed to take about 80 kilo-bits/sec. Assuming each call takes 10 minutes, each call will take 13 kilo-bytes + (80 kilo-bits * 60 * 10) or 48.13 mega-bits. Assuming (to make the math easier) 10 bits = 1 byte, each call will take 4.813 mega-bytes. So, 100,000 calls of 10 minutes (1 million minutes) would consume 481,300 mega-bytes per month or 3,333 calls consuming 16,043 mega-bytes per day. Assuming the busiest hour accounts for about 10% of the daily total, that hour would consist of 333 calls consuming 1,604 mega-bytes. So, my peak would need 4.5 mega-bits per second of bandwidth. Am I in the ballpark? Would anybody venture an estimate of what the peak bandwidth would be if we changed to IAX? With trunking? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth requirements for 1, 000, 000 minutes a month
But who in there right state if mind would use ulaw? Just take them away to the funny farm ha ha ho ho!! :-P gsm, ilbc, g729 etc are a lot better choice. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Steve Edwards wrote: This may expose my ignorance, but here goes :) I've been asked to figure out how much bandwidth would be needed to handle 1,000,000 minutes a month. Here's the environment: ) All calls are received via SIP. ) All calls use the ulaw codec. ) Calls average 10 minutes in duration. ) The busiest hour will account for 10% of the daily total. This is how I'm figuring it... Casual observation shows that SIP setup and teardown takes about 26 UDP packets. Assuming the packets are full (512 bytes) this adds up to about 13 kilo-bytes for each call. I've heard that ulaw (including overhead) is supposed to take about 80 kilo-bits/sec. Assuming each call takes 10 minutes, each call will take 13 kilo-bytes + (80 kilo-bits * 60 * 10) or 48.13 mega-bits. Assuming (to make the math easier) 10 bits = 1 byte, each call will take 4.813 mega-bytes. So, 100,000 calls of 10 minutes (1 million minutes) would consume 481,300 mega-bytes per month or 3,333 calls consuming 16,043 mega-bytes per day. Assuming the busiest hour accounts for about 10% of the daily total, that hour would consist of 333 calls consuming 1,604 mega-bytes. So, my peak would need 4.5 mega-bits per second of bandwidth. Am I in the ballpark? Would anybody venture an estimate of what the peak bandwidth would be if we changed to IAX? With trunking? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0659-0, 12/15/2006 - 12/15/2006 9:47:48 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth requirements for 1, 000, 000 minutes a month
So, my peak would need 4.5 mega-bits per second of bandwidth. Am I in the ballpark? Sounds about right. Or the other way around (if you need to know the peak bandwidth usage): For audio: 1,000,000 minutes/month = 33,000 minutes/day 10% daily usage in 1 hour = 3,300 minutes used 3,300 minutes used in 60 minutes = 55 concurrent calls 80 kbps / 1 call direction * 55 calls = 4.4 Mbps per direction Assuming full duplex audio, you need 4.4 Mbps in + 4.4 Mbps out per call leg. If you route the call so each packet comes in and goes out the network (2 call legs), then double the bandwidth. I guess adding 0.1 Mbps for call setup and tear down is safe. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bandwidth requirements for 1, 000, 000 minutes a month
But who in there right state if mind would use ulaw? Just take them away to the funny farm ha ha ho ho!! :-P I do. Exclusively. I personally don't like the g729 compression (audio quality and license issues) any my customers definitely notice the difference right away and wonder why the quality degraded. I guess I spoiled them with ulaw. So no g729 here. g726-32 on the other hand was acceptable, although the difference is still noticeable. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.2.12 Released
The Asterisk Development Team is pleased to announce the release of Zaptel 1.2.12. This release contains a number of updates: - compatibility with Linux kernel 2.6.19 - bug fixes to the Xorcom Astribank driver (XPP) - various other bug fixes Thanks for supporting Asterisk and Zaptel! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.14 Released
The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.14. This release contains a number of updates: - a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it - various SIP interoperability improvements - memory and dialog leaks in the SIP channel driver - a fix to music-on-hold random mode that was not really random - an improvement to app_voicemail to ensure that the message duration is properly included in email notifications when voicemail messages are forwarded - corrected a segfault issue during reload of the PostgreSQL CDR driver - a change to no longer include a header file that does not exist on Linux kernel 2.6.18 (and caused a problem on Fedora Core 6) - many other bug fixes Thanks for supporting Asterisk and Zaptel! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.4.0-beta3 Released
The Asterisk Development Team is pleased to announce the release of Zaptel 1.4.0-beta3. This release contains a number of updates: - compatibility with Linux kernel 2.6.19 - bug fixes to the Xorcom Astribank driver (XPP) - support for Digium's TE110P Rev C, VPMOCT064 and new revisions of the S110M and S400M FXS modules - various other bug fixes Thanks for supporting Asterisk and Zaptel! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.0-beta4 Released
The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.0-beta4. This release contains a number of updates: - a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it - various SIP interoperability improvements - memory and dialog leaks in the SIP channel driver - a fix to music-on-hold random mode that was not really random - an improvement to app_voicemail to ensure that the message duration is properly included in email notifications when voicemail messages are forwarded - corrected a segfault issue during reload of the PostgreSQL CDR driver - a change to no longer include a header file that does not exist on Linux kernel 2.6.18 (and caused a problem on Fedora Core 6) - logging of dynamic queue member addition and removal in queue_log - a minor redesign of many CLI commands to be more similar to previous Asterisk releases - significant improvements to IMAP storage support for voicemail - a change to the SIP channel to avoid offering formats (codecs) that cannot be transcoded due to lack of available transcoders (along with dynamic activation/deactivation of transcoders) - support for G.722 16KHz (wideband) audio passthrough, recording and playback - support for standard prompts in G.722 format - many other bug fixes Some of the changes in this release are behavior modifications from the last release; please review the UPGRADE.txt file. This will very likely be the last beta release of Asterisk 1.4 before the final release, which is targeted for next Friday. Thanks for supporting Asterisk and Zaptel! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iptables rule help
Google is your friend!! http://www.eweek.com/article2/0,1895,1773983,00.asp http://www.eweek.com/article2/0,1895,1773832,00.asp http://www.eweek.com/article2/0,1895,1772661,00.asp Let us hope SS isn't a communications lawyer The FCC DOES have jurisdiction John Novack Steve Sobol wrote: On Fri, 15 Dec 2006, John Novack wrote: Are you in the US? If so, such blocking is not legal I'd like to see a citation for that. ISPs aren't common carriers and aren't required to carry specific types of traffic. and you should file a complaint with the FCC The FCC regulates common carriers. ISP's aren't. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Commercial Grade Service Provider?
VoicePulse is the absolute worst. You can get additional channels for $25/month but that includes no usage whatsoever. That's almost double what the same capacity WITH MINUTES on a PRI port costs! Any decent provider will be able to give you an unlimited number of channels because you are paying for the usage. If you are paying per channel I would expect some sort of included usage. For example one of our PRI provider's offering boils down to about $12/channel, unlimited regional calling (more than Bell's local calling area) and some 200 minutes of LD calling extra DID cost less than a quarter each, compare that to voicepulse charging you $25/month for jack shit $11/month per DID but no additional usage. You can get 20 DID with them on one account and you get 4 calls at a time your cost is $220/month. you can open however 20 different accounts with one number on each, you pay the SAME $220/month however you get 80 calls at the same time! If you wanted the same arrangement on a single voicepulse account it would cost $620/month However don't do that, with a single account VoicePulse will charge you RANDOM amounts to your credit card, even if they say they will ONLY charge your card in $25 increments, I've asked them countless times to charge other amounts and they say NO impossible, billing system limitation, yada yada but when it comes down to it they can do and will charge your card for a random amount. And you cannot port any telephone number away from them, they have instructed their carrier (Broadview) to not allow any sort of LNP out request. Also any time there is an issue they blame you. And aulthough they sell a VoicePulse Connect! for Asterisk service where Asterisk is a LINUX PROGRAM they insist you run a WINDOWS PROGRAM on the same machine for troubleshooting, when you remind them you are running Windows they tell you to run WINE when you remind them that even Digium recommends you do not run a GUI on the same machine as Asterisk they start to ignore you. On 12/15/06, Paul Connolly [EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tone Issues
Jason: The issue is indeed VoicePulse. Their equipment is not correctly setup and/or capable to recieve DTMF from many sources, one of those is Sprint CDMA mobile phones, they claim the issue is Sprint however Sprint is correctly sending DTMF and every other carrier is able to recieve them. Best regards, Andrew Joakimsen P.S.: Good luck porting VoicePulse numbers, I've just started the headache myself, everytime I ask why our port requests are rejected they just copy and paste their TOS which clearly states that there is a porting fee, when I respond asking what the porting fee is they say something I respond back they copy paste the TOS I respond asking for what is the fee... yada yada yada On 12/15/06, Jason Walker [EMAIL PROTECTED] wrote: I have 1.2.12.1 Voicepulse using IAX I get about 30-40% issues with not having the DTMF tones work. I have 3 questions #1. Voicepulse says they are sending them, Is there some setting I can adjust to make sure my end is working? #2. I have set the Dialplan to play a sound Operator then go to a SIP phone. I would like it to write Caller ID Time to a file I can read and find out exactly how many people are getting to that point. #3. If it is Voicepulses fault. Who else might you suggest for my numbers to be ported to and handle my phone calls Thanks Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International Provider
Carlos Rojas wrote: Anyone know a good carrier of voip for international calls? Please use asterisk-biz list http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Motherboard 3.3V PCI for TE412P
Jesus Mogollon wrote: Hi all Does anyone know of any motherboards with PCI slots that can take the TE412P card? Is there such a MB for Athlon 64 or P4 procs? I have no experience of it, but you could look at the Asus M2N32 WS which has 2 x PCI-X (3.3V) slots. It is a socket AM2 (Athlon64) board. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
