[asterisk-users] R2D2 VOIP Kubuntu 8.4 Ekiga, Ekiga.net voice conference
I'm having a problem getting a good clear output sidnal from Ekiga to a VOIP conference call using the Ekiga.net free conference call system. I'm told that each time I speak, my voice is clear intelligible for about .5 - 2 seconds, but then it starts to be garbled, sounding like the sounds R2D2 makes. I've used 2 or three mic/headsets - two plug into my audio I/O sockets on my laptop, one is a USB headset (but I'm not sure I tested the usb headset properly, though a friend with the exact same usb headset, also on KUbuntu 8.4, like myself, doesn't have the problem.) My voice came through clearly in 1 on 1 conversations to a specific person. I'm told the problem lessens when I turn down the volume of my microphone gain, but I can't recall if that always worked, or just sometimes. One key observation: It worked fine when I was alone, but all the times I've had problems there have been others at the same table as myself who were listening to the conference on laptop speakers - I suspect the problem might be feedback from their speakers to my microphone - if so, perhaps I can solve this by ensuring noone else nearby is in the conference outputting through laptop speakers. -- So, I just want to know if what I've described is a known issue, or if this R2D2 sounding problem has never been noticed before. /or if there is a know solution to this type of problem. Thanks :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very
try do add fromdomain=acme.com/sip.acme.com fromhost=acme.com/sip.acme.com 2009/1/6 Frank Bulk frnk...@iname.com I tried that before, but I just tried it again. Unfortunately, the same thing: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 [ACME] host=172.16.10.40 username=username secret=password type=friend Frank *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Allan Dib *Sent:* Monday, January 05, 2009 9:41 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Try it by IP address instead of hostname as reverse DNS may not be resolving. e.g. host=123.123.123.123 On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk frnk...@iname.com wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which SIP/2.0 401 Unauthorized is issued after the un-authenticated INVITE and SIP/2.0 403 Forbidden after the authenticated INVITE. When I add insecure=very, this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: sip:5551236...@172.16.10.40sip%3a5551236...@172.16.10.40 It isn't very clear (to me) from the success how the insecure=very helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Frank Bulk - iName.com wrote: The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add insecure=very to my Outgoing settings, but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with SIP/2.0 403 Forbidden I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the Incoming Settings panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com sip%3a%2b15552027...@sip.acme.comSIP/2.0 From: sip:5552022...@172.16.10.40 sip%3a5552022...@172.16.10.40 ;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b ba4 To: sip:+15552027...@sip.acme.com sip%3a%2b15552027...@sip.acme.com Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: sip:5552022...@172.16.10.40sip%3a5552022...@172.16.10.40 ;user=phone Privacy: none Remote-Party-ID: sip:5552022...@172.16.10.40sip%3a5552022...@172.16.10.40;user=phone; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: sip:5552022...@172.16.10.40 sip%3a5552022...@172.16.10.40 Authorization: Digest username=username,realm=asterisk,nonce=118af2b0,uri=sip:+15552027020 @ sip.acme.com,response=111e63ec2a1f3ebabefe4f7dae4087a1,algorithm=MD5 Content-Type: application/SDP
[asterisk-users] Problems getting 1.6 to run with user asterisk and group asterisk
I've built SVN-trunk-r167180 and try to start it with: asterisk -f -C /etc/asterisk/asterisk.conf which results in: Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl: Permission denied However, /etc/asterisk/asterisk.conf has: astrundir = /var/run/asterisk runuser = asterisk rungroup = asterisk The directory, user, and group exist. Permissions are fine. The only way I can get it to run is to edit defaults.h and change: #define DEFAULT_RUN_DIR/var/run/asterisk #define DEFAULT_SOCKET /var/run/asterisk/asterisk.ctl #define DEFAULT_PID/var/run/asterisk/asterisk.pid After these changes and recompiling all is fine. It seems that at startup asterisk.conf is being parsed, runuser and rungroup are being set, but astrundir is not overriding the defaults. Am I missing something? Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2D2 VOIP Kubuntu 8.4 Ekiga, Ekiga.net voice conference
Sometimes it's a problem of the timing, do you have this problem with normal call's ? 2009/1/6 john_re john...@fastmail.us I'm having a problem getting a good clear output sidnal from Ekiga to a VOIP conference call using the Ekiga.net free conference call system. I'm told that each time I speak, my voice is clear intelligible for about .5 - 2 seconds, but then it starts to be garbled, sounding like the sounds R2D2 makes. I've used 2 or three mic/headsets - two plug into my audio I/O sockets on my laptop, one is a USB headset (but I'm not sure I tested the usb headset properly, though a friend with the exact same usb headset, also on KUbuntu 8.4, like myself, doesn't have the problem.) My voice came through clearly in 1 on 1 conversations to a specific person. I'm told the problem lessens when I turn down the volume of my microphone gain, but I can't recall if that always worked, or just sometimes. One key observation: It worked fine when I was alone, but all the times I've had problems there have been others at the same table as myself who were listening to the conference on laptop speakers - I suspect the problem might be feedback from their speakers to my microphone - if so, perhaps I can solve this by ensuring noone else nearby is in the conference outputting through laptop speakers. -- So, I just want to know if what I've described is a known issue, or if this R2D2 sounding problem has never been noticed before. /or if there is a know solution to this type of problem. Thanks :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems getting 1.6 to run with user asterisk and group asterisk
On Tue, Jan 06, 2009 at 02:28:53AM -0600, Alejandro Kauffmann wrote: I've built SVN-trunk-r167180 and try to start it with: asterisk -f -C /etc/asterisk/asterisk.conf which results in: Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl: Permission denied However, /etc/asterisk/asterisk.conf has: astrundir = /var/run/asterisk runuser = asterisk rungroup = asterisk Could you please post the complete file? (maybe grep -v '^;') -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
Hi thanks, Each span is connected to a separate Quintum gateway so I took it that each span will need to decide with its own Quintum which side is the source of the timing, I hope my logic is right because with this config I got better results. I also changed headsets to USB which improved by 200% the audio quality for both parties of the call. I am still experiencing with some settings for pci ports kernel at boot time, will update you when I finish it. Thanks again On Tue, Jan 6, 2009 at 3:28 AM, Ex Vito ex.vitor...@gmail.com wrote: IIRC, the second argument in the span lines indicates the timing sync with 1 meaning that this span is master. I'd say it makes no sense to have both of them be masters... I have no current docs / system at hand; give it a check and then, maybe try to have one as master and the other as slave (0 instead of 1, again, IIRC). Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridge 2 calls
I am also interested in establishing a three way conversation using a simple webpage. I wonder if anyone can provide some help on that. On Tue, Jan 6, 2009 at 7:29 AM, amit mehta amit.magn...@gmail.com wrote: Hi Rilawich, I worked recently on it and that is why can give you the idea how i achived it. You can write an PHP script to get the number and name of the customer.You can phpself to the script.Then you can use an API script to use that number to orignate the call.The channel will be used to call the asterisk internal agent and the other line will call the number that was input by the customer and bridge the call. Hope this might help you. Regards, Amit Mehta Cell: +91 9898340962 On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote: Hi all, I want to build a web page for user to input a phone number. Then, the number will input to asterisk and it will makes call. At that moment, asterisk will make another call to a internal ext. Finally asterisk will bridge 2 calls together for conversion. Does asterisk can do it? How? Thanks, Ango ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G.729 VAD issue
Hi, My setup is SIP Call--Asterisk--VSP1 or VSP2 or VSP3 I'm experiencing an interconnect issue with one of the VSP's that seems to have to do with Asterisk not having any VAD control. The error is: NOTICE[10989]: frame.c:203 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end The VSP has switched off silence suppression on their Quitnum device. Any ideas are most welcome. Thanks Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk CLI got freezed!!
Hi All, I am using asterisk 1.4.21 with iaxmodem and hylafax which is sending fax from my system with zap device. I am facing a problem that some times my asterisk CLI got freeze and i am not able to get any information from asterisk. I need to restart the asterisk compulsory to work it again. And because of this my iaxmodems are also getting time out from asterisk. Please provide some help regarding this freeze issue. Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI got freezed!!
Make sure the DNS servers Asterisk is using are not becoming unresponsive or unreachable. Asterisk blocks on DNS requests so if it doesn't get a response it will appear frozen. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agents, Queues and logon/logoff
Tilghman, Thanks for that, that is good information to know. Regards, Pedram On Mon, Jan 5, 2009 at 4:57 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Monday 05 January 2009 16:52:28 Danny Nicholas wrote: Yes, but if you do, you will lose it in a future upgrade (if that matters to you). No, he won't. Our current policy is that while we may deprecate functionality, we will never again remove it (unless the deprecated functionality somehow interferes with our ability to introduce new, better ways of doing things, which I doubt this application will ever do). However, if you're experiencing problems with deprecated functionality, the support you can expect for it will be much lower in priority compared to other, supported functionality. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] username mismatch, have x, digest has y
I have two Asterisks connected using SIP. One is acting as a SIP server, the other as a SIP client. This almost works; but calls from 50607795 are rejected with this error: check_auth: username mismatch, have 50607796, digest has 50607795 On the client I have these accounts configured in sip.conf: register = 50607795:t...@10.10.33.228/50607795 register = 50607796:te...@10.10.33.228/50607796 [50607795] accountcode=mobiltest defaultuser=50607795 type=peer host=10.10.33.228 canreinvite=no insecure=port,invite context=from-inside secret=test fromuser=50607795 trustrpid=yes sendrpid=yes [50607796] accountcode=mobiltest defaultuser=50607796 type=peer host=10.10.33.228 canreinvite=no insecure=port,invite context=from-inside secret=test2 fromuser=50607796 trustrpid=yes sendrpid=yes On the server, these are configured: [50607795] callgroup= pickupgroup= callerid=test 50607795 canreinvite=yes context=do_dial mailbox= secret=test type=peer disallow=all allow=alaw allow=ulaw host=dynamic call-limit=100 dtmfmode=rfc2833 [50607796] callgroup= pickupgroup= callerid=test2 50607796 canreinvite=yes context=do_dial mailbox= secret=test2 type=peer disallow=all allow=alaw allow=ulaw host=dynamic call-limit=100 dtmfmode=rfc2833 Both servers are running Asterisk 1.6.0.1. The problem seems to be that the client registers twice using the same IP address and port. The server then gets confused and doesn't try both sets of credentials -- only the last one mentioned in sip.conf. If I add insecure=invite, then the call is allowed, but the server then believes that it was 50607796 which made the call, when in fact it was 50607795. Not so clever. I can see two ways out of this problem: a) Get the client Asterisk to use a unique port number for each registration to a specific server. This seems like a lot of work for little gain, because devices like Snom M3 exhibit the exact same behaviour, so they would remain broken. b) Make the server Asterisk try to match all the clients with the specific IP-address and port, not just one. This seems like the correct solution. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] enabling silence suppression in asterisk
Hi Friends, Currently i am using the asterisk 1.4.x version. In that i want to enable to silence suppression in the SIP calls. Please tell me the configuration changes to be done. Thanks in advance, balasam. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very
That's a good suggestion, but I tried that and it didn't work. I think you need an '' in-between, so I tried that, too. I also tried adding the IP address of the CS 1500, too, and that didn't help. Frank From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grygoriy Dobrovolskyy Sent: Tuesday, January 06, 2009 2:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very try do add fromdomain=acme.com/sip.acme.com fromhost=acme.com/sip.acme.com 2009/1/6 Frank Bulk frnk...@iname.com I tried that before, but I just tried it again. Unfortunately, the same thing: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 [ACME] host=172.16.10.40 username=username secret=password type=friend Frank From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Allan Dib Sent: Monday, January 05, 2009 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Try it by IP address instead of hostname as reverse DNS may not be resolving. e.g. host=123.123.123.123 On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk frnk...@iname.com wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which SIP/2.0 401 Unauthorized is issued after the un-authenticated INVITE and SIP/2.0 403 Forbidden after the authenticated INVITE. When I add insecure=very, this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: sip:5551236...@172.16.10.40 mailto:sip%3a5551236...@172.16.10.40 It isn't very clear (to me) from the success how the insecure=very helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Frank Bulk - iName.com wrote: The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add insecure=very to my Outgoing settings, but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with SIP/2.0 403 Forbidden I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the Incoming Settings panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com mailto:sip%3a%2b15552027...@sip.acme.com SIP/2.0 From: sip:5552022...@172.16.10.40 mailto:sip%3a5552022...@172.16.10.40 ;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b ba4 To: sip:+15552027...@sip.acme.com mailto:sip%3a%2b15552027...@sip.acme.com Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: sip:5552022...@172.16.10.40 mailto:sip%3a5552022...@172.16.10.40 ;user=phone Privacy: none Remote-Party-ID:
[asterisk-users] Call transfer using agi
Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password of caller. I have an agi to change password and can transfer call to agi, but I do not know how to transfer the call back to agent from agi. So basically how can an agi transfer a call to an extension? Thanks and regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: A2billing Multiple Servers
Hi All Its possible integrate multiple servers using A2billing? what i`m looking for is manage a 1500concurrent calls and have one database, also looking easy scalability lets say i want to handle more calls the only thing i need to do is add another asterisk server. Is this possible? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Asus Eeebox] USB FXO adapter?
Hello I'm contemplating building an Asterisk voice server out of the compact Asus EeeBox: http://www.asus.com/products.aspx?l1=24l2=165 But they're so compact, they don't have a PCI slot to handle an analog phone line. I'd like to minimize footpring and cables: Besides analog/SIP boxes like Linksys (extra cables + transformer), does someone know of a USB adapter that is self-powered and could take an analog line as input, convert voice to SIP, and send packets through the USB port? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] username mismatch, have x, digest has y
Benny Amorsen benny+use...@amorsen.dk writes: I have two Asterisks connected using SIP. One is acting as a SIP server, the other as a SIP client. This almost works; but calls from 50607795 are rejected with this error: check_auth: username mismatch, have 50607796, digest has 50607795 I tried replacing the type=peer with type=friend, but that didn't go very far either: [Jan 6 13:46:02] NOTICE[305]: chan_sip.c:16805 handle_request_invite: Failed to authenticate user 50607795 sip:50607...@x.x.x.x;tag=as589a9bd0 That looked like a 1.6.0 bug, so I tried 1.6.1 beta 4 -- but sip show users is gone in 1.6.1... /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asus Eeebox] USB FXO adapter?
Vincent wrote: Hello I'm contemplating building an Asterisk voice server out of the compact Asus EeeBox: http://www.asus.com/products.aspx?l1=24l2=165 But they're so compact, they don't have a PCI slot to handle an analog phone line. I'd like to minimize footpring and cables: Besides analog/SIP boxes like Linksys (extra cables + transformer), does someone know of a USB adapter that is self-powered and could take an analog line as input, convert voice to SIP, and send packets through the USB port? Thank you. It hasn't been released yet, but this looks like it will do the job: http://wiki.sangoma.com/sangoma-wanpipe-usbfxo People have been reviewing betas since early September so hopefully it will be released soon. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] .call file not updating MySQL CDR's
All; I have implemented an autodialer solution where I create .call files for each number to be dialed. The .call file is very simple in design: Channel: SIP/2405551...@broadvoice-outbound Context: autodial MaxRetries: 5 RetryTime: 600 WaitTime: 60 CallerID: Hildas Cleaning 410555 Extension: 2405551212 Priority: 1 Account: 999 Archive: yes A problem recently started where the MySQL records were not being written (after 6 months of working flawlessly) although all other CDR records were fine. The solution turned out to be where I had to *downgrade* Asterisk from 1.4.22 to 1.4.20.1. Nothing else in the configuration changed. Did I come across a bug, a feature, or what? Did I miss something obvious? Thanks FSD _ Send e-mail faster without improving your typing skills. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_speed_122008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asus Eeebox] USB FXO adapter?
Use xorcom products: www.xorcom.com They provide usb devices for: fox, fxs, bri, pri Best regards, Loic Didelot. On Tue, 2009-01-06 at 10:32 -0500, Dave Fullerton wrote: Vincent wrote: Hello I'm contemplating building an Asterisk voice server out of the compact Asus EeeBox: http://www.asus.com/products.aspx?l1=24l2=165 But they're so compact, they don't have a PCI slot to handle an analog phone line. I'd like to minimize footpring and cables: Besides analog/SIP boxes like Linksys (extra cables + transformer), does someone know of a USB adapter that is self-powered and could take an analog line as input, convert voice to SIP, and send packets through the USB port? Thank you. It hasn't been released yet, but this looks like it will do the job: http://wiki.sangoma.com/sangoma-wanpipe-usbfxo People have been reviewing betas since early September so hopefully it will be released soon. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple CDRs
Greyman-- I'm taking this discussion to the list. Folks, what we are talking about here, is me trying to get a grasp around Greyman's (Aaron's) request for a bare-bones CDR generation that describes just total connect time for channels, stripping out all the details. Who cares about xfer, park, hold, etc.? So in the following is our discussion about what *should* be there, and in what form... So, what I'm thinking, is to spec out two CDR generation modes, one detailed one according to the spec I'm working on, and the other mode will follow these lines... On Tue, 2009-01-06 at 10:37 +, Grey Man wrote: On Mon, Jan 5, 2009 at 6:42 PM, Steve Murphy m...@digium.com wrote: I **think** I have a handle on it... Basically, for each channel that did anything, no matter what, you'd like a single CDR for that channel that would record the time from when it first activated to the time it hung up. I'd have to assume the first answer time would be recorded in the CDR, in case multiple answer times might apply (for incoming calls, it would be the time the pbx 'answered' the incoming call; for outgoing calls it would be the time the other end answered the call... right? Hi murf, To my mind only hangups should generate CDRs and nothing else should. When you say for each channel that did anything I'd like a CDR I'm not to sure about that, if you mean to generate a CDR for every type of channel that is ever hungup then the answer is yes. If you mean to generate a CDR on non-hangup channel events then the answer is no. OK so, if A calls B, B parks A, A's park expires, B is rung, B answers, B xfers A to C, they hang up, we should have a CDR for A's time, with the start time being the time the PBX created the channel for A; the answer time would be (if A is an incoming call) when the PBX answered the incoming call and maybe started giving A the IVR experience, and (if A is an extension), when B answered the call. The end time would be when A was hung up. A CDR for B would be generated? with his answer time when he picked up the phone to answer the incoming call from A? and an end time when he parked A? Another CDR for B would be produced he answered the callback from the PBX for the expired session with A, and end when he got hung up xferring the call to C? Another CDR for C would be produced to record C's conversation with A, start when his phone started ringing, answer when he answered, and end when he hung up? Am I on the right track? I don't use Parking myself so my understanding may be slightly off but from what I do understand of Parking the CDRs would not be generated quite how you describe. The main point is that Parking a call should not generate a CDR as the Park operation has not necessarily ended a call. Parking a call will hang you up, in most normal cases. This includes calling the Park() app, bxfer to the parking exten, and using the one-touch parking features. But, if some strange combo of events allows someone to park without a hangup, then I'd agree, no CDR should be generated. In your description I think the CDRs should be: 1 The call from A to the PBX, start time when B answers, end time when the A-C call is hungup, Can't do this; it would be inaccurate; start time is when A either picks up the phone (if dahdi exten), or when A submits an invite (if sip exten), or when an incoming call (via sip invite, or dahdi fxo i/f) arrives at the pbx. As to Answers, we have to start getting pedantic; if A is an exten, then the first answer will be when B answers. But if A is an incoming call via, say a dahdi fxo interface, or an incoming sip invite, then an s exten is going to get run, and usually the PBX runs the answer() app, and this will usually be the first answer. Now, I can use heuristics to override this first answer if a dial occurs, but... if multiple dials occur, this heuristic would tend to record the last answer; if we override only Answer() in the 's' exten, then we would only record the first answer... is this more like it? Oh, and BUSY/NO ANSWER/FAIL for a non-s exten, would also override an ANSWER on exten s, BTW... And, would it be proper to include all dial attempts? My guess is that you would *NOT* want to see any dial attempts in this mode. Well, at least, in this particular case, if A *tries* to dial B, but B doesn't answer, then since A is a live channel, we would record it's life in the system. When A hangs up, we would see the NO ANSWER disposition, and the destination of B, right? If A tried to dial a group, and nobody answered, the destination would be a random member of that group, the args to the Dial command would record the other members, usually. 2. The call from the PBX to B, start time when B answers, end time when the call to B is hungup once the blind transfer of A to C is initiated, The start time will be when B is first dialed; The answer time when he answers. I
Re: [asterisk-users] [SPAM] enabling silence suppression in asterisk
bala krishnan wrote: Hi Friends, Currently i am using the asterisk 1.4.x version. In that i want to enable to silence suppression in the SIP calls. Please tell me the configuration changes to be done. Thanks in advance, balasam. Enabling silence suppression is a bad thing. Asterisk and sip phone will think that the other party has dropped off and will randomly drop calls on you because of a lack of traffic from the other party. Lyle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as MGCP client
Bob Pierce wrote: Has there been any work done on using Asterisk as an MGCP client? Nope :-( Still a no go. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very
Frank Bulk wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend Your problem is you are trying to do authenticate by host and by username at the same time. That does not work in asterisk. You should be seeing a Warning message in the console saying something like: check_auth: username mismatch, have ACME, digest has username That means you already matched to sip.conf entry ACME, but the digest has a different username, so it fails. You can fix it by setting the paramters in the CS1500 to have the username = ACME. That way the digest will come in as: Digest username=ACME ...bla bla bla Andres http://www.telesip.net I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which SIP/2.0 401 Unauthorized is issued after the un-authenticated INVITE and SIP/2.0 403 Forbidden after the authenticated INVITE. When I add insecure=very, this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: sip:5551236...@172.16.10.40 It isn't very clear (to me) from the success how the insecure=very helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Frank Bulk - iName.com wrote: The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add insecure=very to my Outgoing settings, but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with SIP/2.0 403 Forbidden I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the Incoming Settings panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: sip:5552022...@172.16.10.40;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b ba4 To: sip:+15552027...@sip.acme.com Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: sip:5552022...@172.16.10.40;user=phone Privacy: none Remote-Party-ID: sip:5552022...@172.16.10.40;user=phone; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: sip:5552022...@172.16.10.40 Authorization: Digest username=username,realm=asterisk,nonce=118af2b0,uri=sip:+15552027020 @ sip.acme.com,response=111e63ec2a1f3ebabefe4f7dae4087a1,algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782 2973921782 IN IP4 172.16.10.65 s=SIP Call c=IN IP4 172.16.10.65 t=0 0 m=audio 36224 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very
You're the miracle worker! Thanks! Frank From: Andres [mailto:and...@telesip.net] Sent: Tuesday, January 06, 2009 11:19 AM To: Frank Bulk Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Frank Bulk wrote: This is what I have in my configuration now: [ACME] host=sip.acme.com username=username secret=password type=friend Your problem is you are trying to do authenticate by host and by username at the same time. That does not work in asterisk. You should be seeing a Warning message in the console saying something like: check_auth: username mismatch, have ACME, digest has username That means you already matched to sip.conf entry ACME, but the digest has a different username, so it fails. You can fix it by setting the paramters in the CS1500 to have the username = ACME. That way the digest will come in as: Digest username=ACME ...bla bla bla Andres http://www.telesip.net I've done a SIP debug before, but I've done it again with the above configuration: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 after which SIP/2.0 401 Unauthorized is issued after the un-authenticated INVITE and SIP/2.0 403 Forbidden after the authenticated INVITE. When I add insecure=very, this is what the SIP debug shows: No user '5551236049' in SIP users list Found peer 'ACME' for '5551236049' from 172.16.10.40:5060 Found RTP audio format 0 Peer audio RTP is at port 172.16.10.65:36272 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.10.65:36272 Looking for +15552127020 in from-sip-external (domain sip.acme.com) list_route: hop: sip:5551236...@172.16.10.40 sip:5551236...@172.16.10.40 It isn't very clear (to me) from the success how the insecure=very helps. Frank -Original Message- From: Andres [mailto:and...@telesip.net] Sent: Monday, January 05, 2009 7:43 PM To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very Frank Bulk - iName.com wrote: The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add insecure=very to my Outgoing settings, but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with SIP/2.0 403 Forbidden I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the Incoming Settings panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Do a sip debug on the asterisk console and see if it is actually is matching one of your sip.conf entries during an invite from the CS1500. Look for a line that says something like 'Found Peerbla bla bla'. If you dont see that line, then you are not even adding the correct sip.conf entry to match the invite from the CS1500. Andres http://www.telesip.net Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: sip:5552022...@172.16.10.40 sip:5552022...@172.16.10.40;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d b ba4 To: sip:+15552027...@sip.acme.com sip:+15552027...@sip.acme.com Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: sip:5552022...@172.16.10.40;user=phone sip:5552022...@172.16.10.40;user=phone Privacy: none Remote-Party-ID: sip:5552022...@172.16.10.40;user=phone sip:5552022...@172.16.10.40;user=phone; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: sip:5552022...@172.16.10.40 sip:5552022...@172.16.10.40 Authorization: Digest username=username,realm=asterisk,nonce=118af2b0,uri=sip:+15552027020 @ sip.acme.com,response=111e63ec2a1f3ebabefe4f7dae4087a1,algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782
[asterisk-users] Asterisk Generating NetworkOOO (ISDN Cause Code 38)
I have a legacy ISDN PBX (Network Alchemy Argent Office) connected to Span 2 of a Digium Wildcard TE205P. Recently Calls from this PBX have been failing with ISDN Cause Code 38 (Network Out of Order!). The problem seems to be getting worse and is now effecting more calls than not (although this could just be because I'm aware of it). Once the ISDN PBX has decided the Network's Out of Order and torn down the call, the Asterisk box still has the channels bridged and up. Can anyone suggest a course of action I might take to getting this sorted? Do I need to change something in zapata.conf? (included below). Here's the log from the ISDN PBX showing the call failing: 2592754mS CMCallEvt:v=1129 State, new=Ringing old=Dialled,0,0,BState 2592800mS CMCallEvt:v=1129 State, new=Ringing old=Dialled,0,0,Astate 2595647mS ISDNL3Evt: v=0 stacknum=0 State, new=Active, old=Delivered id=1130 2595657mS CMLineRx: v=1 CMConnect Line: type=Q931Line 1 Call: lid=0 id=1130 in=0 BChan: slot=0 chan=16 2595657mS CMCallEvt:v=1129 State, new=Connected old=Ringing,0,0,BState 2595729mS CMCallEvt:v=1129 State, new=Connected old=Ringing,0,0,Astate 2596653mS ISDNL3Evt: v=0 p1=0,p2=1001,p3=5,p4=0,s1= 2596654mS ISDNL3Evt: v=0 stacknum=0 State, new=NullState, old=Active id=1130 2596659mS ISDNL3Evt: v=0 p1=0,p2=1000,p3=0,p4=0,s1= 2596665mS CMLineRx: v=1 CMReleaseComp Line: type=Q931Line 1 Call: lid=0 id=1130 in=0 Cause=38, NetworkOOO 259mS CMCallEvt:v=1129 State, new=Idle old=Connected,0,0,Astate 2596667mS CMCallEvt:v=1129 State, new=Idle old=Connected,0,0,BState 2596670mS CALL:2009/01/0612:04,00:00:01,002,1780471800,O,0173323,0173323,Modem0,,,0 but Asterisk thinks the call is up and bridged: asterisk*CLI show channels Channel Location State Application(Data) Zap/1-1 (None) Up Bridged Call(Zap/46-1) Zap/46-1 0173323x...@alchemy: Up Dial(Zap/g1/0173323|) 2 active channels 1 active calls asterisk*CLI Here's the relevant bit from zaptel.conf: # Second port span=2,0,1,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 # # Global data loadzone=uk defaultzone=uk ...and here's the bit from my zapata.conf for Span 2. ; ; Network Alchemy ; group = 2 switchtype=euroisdn context = alchemy usecallerid=yes signalling = pri_net resetinterval=1 callerid=asreceived pridialplan=unknown prilocaldialplan=unknown useincomingcalleridonzaptransfer=yes channel = 32-46 Can anyone help? -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com 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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asus Eeebox] USB FXO adapter?
On Tue, 06 Jan 2009 16:51:40 +0100, Loic Didelot ldide...@mixvoip.com wrote: Use xorcom products: www.xorcom.com They provide usb devices for: fox, fxs, bri, pri Thanks but apparently, they don't have single-line USB devices, just a whole bank: www.xorcom.com/telephony-interfaces/telephony-interfaces.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call file not updating MySQL CDR's
You can follow this issue here: http://bugs.digium.com/view.php?id=14167 Your best bet in the future is to check the bug tracker for any issues you may have to see if it has already been reported. Thanks! Leif Madsen. cbbs...@hotmail.com wrote: A problem recently started where the MySQL records were not being written (after 6 months of working flawlessly) although all other CDR records were fine. The solution turned out to be where I had to *downgrade* Asterisk from 1.4.22 to 1.4.20.1. Nothing else in the configuration changed. Did I come across a bug, a feature, or what? Did I miss something obvious? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very
After many hours of fiddling around, Andres gave me the final piece. For those looking to implement SIP Trunks on a CS-1500 with Asterisk, here are the pieces: Diagram: CS-1500 -- customer PBX (172.16.10.40)(172.16.10.195) HOST: should be the DNS name assigned to the CS-1500's SIP interface. e.g. sip.acme.com NUSR: user name used for the CS 1500 to login into the customer PBX. Needs to match up FreePBX's Trunk Name. For those who use the CLI, this section in sip.conf is encased in square brackets. i.e. [customername] NPSW: password used for the CS 1500 to login into the customer PBX. Needs to match up with the secret= line. i.e. secret=password IP: IP address of the customer PBX. i.e. 172.16.10.195 LUSR: user name used for the customer PBX to login into the CS 1500. Needs to match up with the username= line. i.e. username=customername LPSW: password used for the customer PBX to login into the CS 1500. Needs to match up with the secret= line. i.e. secret=password. For simplicity we made NUSR/LUSR the same and NPSW/LPSW the same. Since you need to define a trunk per customer, it makes the most sense and it easiest to support and implement. Here's what you need to add to Asterisk's sip.conf (yes, just those few lines!) [customername] host=sip.acme.com type=friend username=customername secret=password And the CS-1500 output: TYP TG NUM 1234 TGTP 2WAY TGNM SIP MG NO SIGT SIP STSI 0 HNPA 555 RC 0 RTP 0 TRNL PRFX PRFX 24 APFX NONE TRFC NONE 4XCD YES ACKA NO TYPC NOCO NXX UNKN LATA 000 CMCT NO TGID NONE SIT NO CNAR NO LRN NONE TNDM NO LDAT NO TRFC NONE EOAT NO ATIC NO CMCO NO TGMU NO HOST sip.acme.com NUSR customername NPSW password IP 172.16.10.195 PORT 5060 PROT UDP T38F NO AUTH YES LUSR customername LPSW password CLIM 7 CPBY 0 Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Bulk - iName.com Sent: Monday, January 05, 2009 6:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming side of SIP trunk does not work unless I add insecure=very The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add insecure=very to my Outgoing settings, but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with SIP/2.0 403 Forbidden I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the Incoming Settings panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: sip:5552022...@172.16.10.40;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db ba4 To: sip:+15552027...@sip.acme.com Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: sip:5552022...@172.16.10.40;user=phone Privacy: none Remote-Party-ID: sip:5552022...@172.16.10.40;user=phone; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: sip:5552022...@172.16.10.40 Authorization: Digest username=username,realm=asterisk,nonce=118af2b0,uri=sip:+15552027020@ sip.acme.com,response=111e63ec2a1f3ebabefe4f7dae4087a1,algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782 2973921782 IN IP4 172.16.10.65 s=SIP Call c=IN IP4 172.16.10.65 t=0 0 m=audio 36224 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] any SIP client for BlackBerry?
Hi You all, Does anyone know any SIP client for BlackBerry? thank you -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple CDRs
Steve Murphy wrote: So, what I'm thinking, is to spec out two CDR generation modes, one detailed one according to the spec I'm working on, and the other mode will follow these lines... Hmmm... I'm liking this idea so far. On Tue, 2009-01-06 at 10:37 +, Grey Man wrote: so, if A calls B, B parks A, CDR generated (hangup event due to attd-xfer) A's park expires, B is rung, B answers, B xfers A to C, they hang up, CDR generated (hangup event) we should have a CDR for A's time, with the start time being the time the PBX created the channel for A; the answer time would be (if A is an incoming call) when the PBX answered the incoming call and maybe started giving A the IVR experience, and (if A is an extension), when B answered the call. The end time would be when A was hung up. In the above scenario, would there be a CDR for when B was called (start time when B answers, or some other event in the dialplan that causes an answer, i.e. Answer(), Playback(), etc...), and then and then hungup? The hangup would be when B attd-xfer A to the parking lot, and B was then hung up. A CDR for B would be generated? with his answer time when he picked up the phone to answer the incoming call from A? and an end time when he parked A? Aha... it seems we agree :) Another CDR for B would be produced he answered the callback from the PBX for the expired session with A, and end when he got hung up xferring the call to C? Agreed. Another CDR for C would be produced to record C's conversation with A, start when his phone started ringing, answer when he answered, and end when he hung up? Also agreed. Am I on the right track? Based on the premise that a CDR would be generated whenever a channel was hung up, then yes, it appears I'm in agreement with you. I don't use Parking myself so my understanding may be slightly off but from what I do understand of Parking the CDRs would not be generated quite how you describe. The main point is that Parking a call should not generate a CDR as the Park operation has not necessarily ended a call. Parking a call will hang you up, in most normal cases. This includes calling the Park() app, bxfer to the parking exten, and using the one-touch parking features. But, if some strange combo of events allows someone to park without a hangup, then I'd agree, no CDR should be generated. I'm in agreement with how Murf has described the above scenario. If you're going to keep it simple(tm) and generate a CDR whenever a channel is hung up, then what Murf has outlined would generate the CDRs as described. In your description I think the CDRs should be: 1 The call from A to the PBX, start time when B answers, end time when the A-C call is hungup, Can't do this; it would be inaccurate; start time is when A either picks up the phone (if dahdi exten), or when A submits an invite (if sip exten), or when an incoming call (via sip invite, or dahdi fxo i/f) arrives at the pbx. If at all possible, it would be nice if you could build the A-C hangup time, i.e. when call enters the PBX, and when the call is disconnected from the PBX. Ideally you could get a less fine grained picture of a single channels life in the PBX, time-wise. As to Answers, we have to start getting pedantic; if A is an exten, then the first answer will be when B answers. But if A is an incoming call via, say a dahdi fxo interface, or an incoming sip invite, then an s exten is going to get run, and usually the PBX runs the answer() app (or a Background(silence/1...) or Playback(silence/1...) which would also answer the channel) and this will usually be the first answer. Now, I can use heuristics to override this first answer if a dial occurs, but... if multiple dials occur, this heuristic would tend to record the last answer; if we override only Answer() in the 's' exten, then we would only record the first answer... is this more like it? Simple CDRs should be as simple as possible. No heuristics should be done automatically. Perhaps there would be enough information in the CDRs to do this after the fact? Oh, and BUSY/NO ANSWER/FAIL for a non-s exten, would also override an ANSWER on exten s, BTW... And, would it be proper to include all dial attempts? My guess is that you would *NOT* want to see any dial attempts in this mode. Well, at least, in this particular case, if A *tries* to dial B, but B doesn't answer, then since A is a live channel, we would record it's life in the system. When A hangs up, we would see the NO ANSWER disposition, and the destination of B, right? If A tried to dial a group, and nobody answered, the destination would be a random member of that group, the args to the Dial command would record the other members, usually. I'm in agreement with this. A is a live channel. It was processed (I specifically don't say 'answered' here), and then terminated. The disposition should tell me something about the attempted
Re: [asterisk-users] any SIP client for BlackBerry?
Take a look at TRUPHONE @ truphone.com Eric On Tue, Jan 6, 2009 at 1:33 PM, TianLun Song stl...@gmail.com wrote: Hi You all, Does anyone know any SIP client for BlackBerry? thank you -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asus Eeebox] USB FXO adapter?
Vincent wrote: Hello I'm contemplating building an Asterisk voice server out of the compact Asus EeeBox: http://www.asus.com/products.aspx?l1=24l2=165 But they're so compact, they don't have a PCI slot to handle an analog phone line. I'd like to minimize footpring and cables: Besides analog/SIP boxes like Linksys (extra cables + transformer), does someone know of a USB adapter that is self-powered and could take an analog line as input, convert voice to SIP, and send packets through the USB port? This looks promising: http://blog.voipsupply.com/asterisk-hardware/first-look-sangoma-u100-usb-fxo-interface-device http://wiki.sangoma.com/sangoma-wanpipe-usbfxo Andres http://www.telesip.net Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any SIP client for BlackBerry?
Thank you, This one looks much better. Is it able to register with Asterisk instead of sign up a plan with Truphone? thank you On Tue, Jan 6, 2009 at 2:02 PM, Eric Moniz emoni...@gmail.com wrote: Take a look at TRUPHONE @ truphone.com Eric On Tue, Jan 6, 2009 at 1:33 PM, TianLun Song stl...@gmail.com wrote: Hi You all, Does anyone know any SIP client for BlackBerry? thank you -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue
Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue
record it... 2009/1/6 Mateusz Pawlowski js+aster...@yllq.net js%2baster...@yllq.net Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue
Why not just make a moh file of a ring-tone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mateusz Pawlowski Sent: Tuesday, January 06, 2009 1:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue
Mateusz Pawlowski wrote: Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. I would suggest recording a ringing sound and play it back as MOH. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue
Check out the r parameter, http://www.voip-info.org/wiki-Asterisk+cmd+Queue Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, January 06, 2009 16:08 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queue Why not just make a moh file of a ring-tone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mateusz Pawlowski Sent: Tuesday, January 06, 2009 1:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue
Mateusz Pawlowski wrote: Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz You can pass the 'r' option to the Queue application for this purpose. As an example: exten = 5000,1,Queue(MyQueue,r) Note that if you are using an Asterisk version prior to 1.6.0, this will have the side-effect of not playing any sort of configured sounds to the caller while he is waiting, e.g. hold time or position announcements. He will hear nothing but ringing until someone answers. Mark! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel variable to identify the calling SIP peer
since 1.4 you can also use setvar=foo=bar in sip.conf when configuring the peer. Then the channel variable foo is automatically set to bar for calls initiated by this peer. regards klaus Philipp Kempgen wrote: Grey Man schrieb: On Wed, Nov 26, 2008 at 11:47 AM, Richard Brady rnbr...@gmail.com wrote: Hi folks I'm not sure what I am missing but I cannot find a predefined channel variable to identify the SIP peer/user which has initiated a call and established the channel. The one option is to extract it from the CHANNEL variable, but that is fraught with difficulties. Is there another variable I don't know about or another way to do this? In 1.2 and 1.4 I don't believe there is any other way. Parsing the username from the channel name is what we ended up having to do! Since 1.6 there is CHANNEL(peername). Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 and LUA
Hello all. I'm playing with LUA and I can't see a way to reload 'extensions.lua' after a change, except by restarting Asterisk. Any clue? Thanks. - Dominique Dartois ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Web-driven SIP call thru Asterisk IPBX
Just let you know that the SIP webphone service is also reachable on
doddling.com
You can pre fill it with your SIP settings:
http://doddling.com/endoddle.jsp?sipserver=MyServersiprealm=Realmcallto=Phoneusername=Userprovider=Namehide=y
Paulo
Doddle WebPhone
doddling.com
From: Paulo Vicentini pvicentin...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Monday, December 22, 2008 5:17:59 PM
Subject: Web-driven SIP call thru Asterisk IPBX
Hi,
I think that the web-driven SIP Phone (free) doddle (beta version) can be
useful with your Asterisk applications.
You can pre-fill it with your sip settings (Asterisk host name or IP / realm /
sip user), you just need to setup the HTML link as that: (Attached is the HTML
page example)
/**/
simple HTML code example:
/*/
html
head
script type=text/javascript
function webcall_win(sip,realm,phone,user,serviceName)
{
//You can have your ajax code here communicating with your site...
//XMLHttpRequest...
var URL =
http://doddle.com.br/endoddle.jsp?sipserver=+sip+siprealm=+realm+callto=+phone+username=+user+provider=+serviceName;
window.open(URL,MyWindow)
}
/script
/head
body
h3Your Asterisk Applications web site.../h3
pUse Asterisk to call right now!
a
href=javascript:webcall_win('asteriskIP','asterisk','123456','myuser','myServiceName');uWeb-driven
Call/u/a
/body
/html
/*/
Thus your Asterisk sip users are ready to call from web page with your Asterisk
server.
PS: Asterisk’s default realm: asterisk
sip.conf:
[general]
realm = your_realm_here / default is asterisk
Address: www.doddle.com.br
Paulo
-Inline Attachment Follows-
My Asterisk Applications web site here...
Use our Asterisk to call right now! Web-driven Call
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[asterisk-users] If you use Realtime Extensions... READ THIS...
One of the frequently asked-for features in pbx_realtime is the ability not to have to have an extensions.conf, because you want realtime to auto-register its contexts. There is now such a patch out there, for testing. The faster that people test it and give feedback, the sooner it can make its way into Asterisk. http://bugs.digium.com/view.php?id=14158 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] enabling silence suppression in asterisk
I've found silence suppression is terrible in practice and not worth the bandwidth saving... You wil have major audio clipping problems with it. On Wed, Jan 7, 2009 at 12:18 AM, bala krishnan mbk_b...@rediffmail.comwrote: Hi Friends, Currently i am using the asterisk 1.4.x version. In that i want to enable to silence suppression in the SIP calls. Please tell me the configuration changes to be done. Thanks in advance, balasam. [image: Ishare]http://adworks.rediff.com/cgi-bin/AdWorks/click.cgi/www.rediff.com/signature-home.htm/1050715...@middle5/2652905_2645144/2648371/1?PARTNER=3OAS_QUERY=null ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Personal Development Without The Silly Stuff: http://AllanDib.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Simple CDRs
On Tue, Jan 6, 2009 at 3:53 PM, Steve Murphy m...@digium.com wrote: Can't do this; it would be inaccurate; start time is when A either picks up the phone (if dahdi exten), or when A submits an invite (if sip exten), or when an incoming call (via sip invite, or dahdi fxo i/f) arrives at the pbx. I understand what you are saying about start time and I think your approach is correct. I was a bit loose in the use of start and answer time but I agree with you. The answer time is absolutely critical but start time is also required as it can be used for things like identifying how long a user waited for a call to answer. As to Answers, we have to start getting pedantic; if A is an exten, then the first answer will be when B answers. But if A is an incoming call via, say a dahdi fxo interface, or an incoming sip invite, then an s exten is going to get run, and usually the PBX runs the answer() app, and this will usually be the first answer. Now, I can use heuristics to override this first answer if a dial occurs, but... if multiple dials occur, this heuristic would tend to record the last answer; if we override only Answer() in the 's' exten, then we would only record the first answer... is this more like it? That sounds a bit dangerous to me. If you go down the path of setting the answer time based on dial plan applications or events you'll need to understand and modify every dial plan application that can answer a call. To me it would seem a lot simpler to do the override/modification in each channel or even better even lower in ast_channel. A channel has to have a very clearly defined definition of answer and hangup whereas dial plan applications don't. Oh, and BUSY/NO ANSWER/FAIL for a non-s exten, would also override an ANSWER on exten s, BTW... And, would it be proper to include all dial attempts? My guess is that you would *NOT* want to see any dial attempts in this mode. Well, at least, in this particular case, if A *tries* to dial B, but B doesn't answer, then since A is a live channel, we would record it's life in the system. When A hangs up, we would see the NO ANSWER disposition, and the destination of B, right? If A tried to dial a group, and nobody answered, the destination would be a random member of that group, the args to the Dial command would record the other members, usually. I liked you previous approach where all call attempts were recorded and there was a config option to opt out of CDRs for non-answered calls for people that didn't want them. When the Dial command specifies multiple destinations then there should be one CDR for each destination that is dialled irrespective of whether it is answered or not. A disposition of something like CANCELLED could be set for the dial legs that Asterisk cancels after the first one is answered. As an example consider the standard call scenario where a user calls into Asterisk and the dialplan forwards the call to 3 destinations: User A -- Asterisk: Dial(SIP/xSIP/ySIP/z) -- SIP/y answers That should generate 4 CDRs: 1. A to Asterisk which is answered, 2. Asterisk to X which is cancelled, 3. Asterisk to Y which is answered, 4. Asterisk ti Z which is cancelled. For people setting the no unanswered call CDRs the 2 and 4th CDRs would not be generated. I notice that you group the two B CDRs I described into a single entity, but in doing so, you violate your own rule; when B parked A, B was hung up. (He could easily dialed party D, eg, and had a conversation and hung up while A was parked!) According to the rules, there should be two CDR's for B, right? I think this was just me not being that familiar with parking. If the example had been a transfer I would have been on the ball :-). OK, gotcha. Now, let's talk about the fields in current/future CDRs, and see which you consider relevant? The core fields I would put into the Asterisk CDRs are: - uniqueid: A GUID/UUID that cannot be changed and is critical for billing, - calldirection: 0 for a call Asterisk receives and 1 for a call Asterisk initiates, - accountcode (user modifiable) - clid: The channel identifier of the call originator equivalent to the A number on a traditional telco CDR, - dst: The destination of the call equivalent to the B number on a traditional telco CDR, - starttime: The time Asterisk first receives or initiates a call, - progresstime: The time Asterisk first receives or generates a progress indication, - answertime: The time a call is answered, - endtime: The time a call is hungup or cancelled, - duration: endtime - starttime, - billsec: endtime - answertime, - disposition: ANSWERED, TIMEOUT, CANCELLED, HUNGUP and maybe others, - userfield (user modifiable): General purpose field for any custom CDR info needed by Asterisk users. Some extra fields that I think would also be very (if not very very) useful to people: - remoteip: The remote IP address of the call where relevant. For an incoming call the originator's IP address, for an
Re: [asterisk-users] Problems getting 1.6 to run with user asterisk and group asterisk
Tzafrir Cohen wrote: On Tue, Jan 06, 2009 at 02:28:53AM -0600, Alejandro Kauffmann wrote: I've built SVN-trunk-r167180 and try to start it with: asterisk -f -C /etc/asterisk/asterisk.conf which results in: Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl: Permission denied However, /etc/asterisk/asterisk.conf has: astrundir = /var/run/asterisk runuser = asterisk rungroup = asterisk Could you please post the complete file? (maybe grep -v '^;') Tzafrir here is the output you requested: [directories](!) ; remove the (!) to enable this astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk [options] runuser = asterisk ; The user to run as rungroup = asterisk ; The group to run as documentation_language = en_US ; Set the Language you want Documentation displayed in. Value is in the same format as locale names [compat] pbx_realtime=1.6 res_agi=1.6 app_set=1.6 Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems getting 1.6 to run with user asterisk and group asterisk
On Tue, Jan 06, 2009 at 09:39:36PM -0600, Alejandro Kauffmann wrote: Tzafrir Cohen wrote: On Tue, Jan 06, 2009 at 02:28:53AM -0600, Alejandro Kauffmann wrote: I've built SVN-trunk-r167180 and try to start it with: asterisk -f -C /etc/asterisk/asterisk.conf which results in: Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl: Permission denied However, /etc/asterisk/asterisk.conf has: astrundir = /var/run/asterisk runuser = asterisk rungroup = asterisk Could you please post the complete file? (maybe grep -v '^;') Tzafrir here is the output you requested: [directories](!) ; remove the (!) to enable this With the '(!)' this section has no effect. astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] \iaxclient-2.0.2 compile problem
Hi, I had downlaoded iaxclient-2.0.2 and complie project *\iaxclient-2.0.2\contrib\win\vs2005* ** It gives many83 fatal and file missing error of file missing Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c 40 Error 2 fatal error C1083: Cannot open source file: '..\..\..\..\libtheora\lib\toplevel.c': No such file or directory c1 Error 3 fatal error C1083: Cannot open source file: '..\..\..\..\libtheora\lib\scan.c': No such file or directory c1 * .* * .* * .* Error 80 fatal error C1083: Cannot open include file: 'theora/theora.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\codec_theora.c 72 Error 81 fatal error C1083: Cannot open include file: 'speex/speex.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\codec_speex.h 15 Error 82 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_common\portmixer.h 47 Error 83 fatal error C1083: Cannot open include file: 'speex/speex.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\codec_speex.h 15 i dont know from where i got missing file Please help MK * * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI got freezed!!
HI, Thanks for your reply, But we have not setup DNS servers in asterisk. Asterisk is not getting any DNS requests. Please provide help regarding this. Thanks, Max Alex Voip Developer On Tue, Jan 6, 2009 at 4:10 PM, Grey Man greymanv...@gmail.com wrote: Make sure the DNS servers Asterisk is using are not becoming unresponsive or unreachable. Asterisk blocks on DNS requests so if it doesn't get a response it will appear frozen. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI got freezed!!
Doesn't matter if you have set it up or not Asterisk needs DNS. I haven't checked the code but I think it even does reverse lookups on IP addresses. If you haven't got a reliable DNS server available for Asterisk I suspect you're always going to get issues. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
