[asterisk-users] GSM to SIP Adapter
Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel (one SIM card). any suggestions? Tarek Sawah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM to SIP Adapter
Thank you for the reply, actually we are looking for something like the followinghttp://www.ebay.com/itm/GSM1SIP-GSM-over-IP-GoIP-SIP-Quad-Bands-voip-gateway-Quad-band-1XGSM-GoIP-VoIP-/181075100268how ever our requirement are a bit wire like SMS in addition to Call capability. Tarek Sawah From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Fri, 11 Oct 2013 15:33:36 +0100 Subject: Re: [asterisk-users] GSM to SIP Adapter On Friday 11 October 2013, Tarek Sawah wrote: Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel (one SIM card). any suggestions? Tarek Sawah We've been using OpenVox G400P cards (PCI; there is also a G400E, which is PCI express for newer motherboards). Sends and receives text messages, and makes and answers phone calls. Accepts up to four RF modules, each of which accepts one SIM card. If you only need text message functionality (not voice calls), then almost any old mobile phone with a USB or RS232 cable can be used as a GSM modem -- and you probably have one lying in a drawer. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SET SIP_CODEC and Video issues
Greetings List.
I Have a small test server and i'm facing a small issue.
i have setup two SIP PEERS and they are able to do Video calls.
now I'm testing SET SIP_CODEC in a dial plan and when ever i'm setting the
codec .. the inbound (=first) leg stops receiving or sending video and SIP SHOW
CHANNELS shows only the Codec i set in the dialplan.
is it possible to avoid this problem?
Asterisk version
1.8.11.0
SIP.CONF
===
[TK1000]
type=friend
secret=0jCiOdT81P
videosupport=yes
qualify=yes
host=dynamic
dtmfmode=rfc2833
context=DER-TEST
canreinvite=yes
disallow=all
allow=ulaw,alaw,gsm,h263,h263p
[TK1000]
type=friend
secret=0jCiOdT81P
videosupport=yes
qualify=yes
host=dynamic
dtmfmode=rfc2833
context=DER-TEST
canreinvite=yes
disallow=all
allow=ulaw,alaw,gsm,h263,h263p
EXTENSIONS.CONF
[DER-TEST]
;exten = _.,1,NoCDR()
exten = _.,1,Set(SIP_CODEC=alaw)
exten = _.,2,Set(SIP_CODEC_OUTBOUND=gsm)
;exten = _.,2,Set(SIP_CODEC_INBOUND=gsm)
exten = _.,n,DIAL(SIP/TK${EXTEN})
exten = h,1,Hangup()
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
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[asterisk-users] Best PBX for Call Centers?
Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk. i'm not an expert with Sippy so i'm looking for a piece of an advise here.. if i'm doing an Asterisk Vs Sippy comparison. can anyone help? Regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems during calls
Aksel, i faced a similar issue with remote sip extensions. and seems to be happening due to internet problems. one way audio that is .. one of the parties (on site) stops hearing the other party. and it happens with one extension at a random timing and random extension.. and if all extensions are on the same internet link it doesnt' happen to all of them at once.. only one of them. i suggest trying to change ISP for testing. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: ak...@abacus-it.no To: asterisk-users@lists.digium.com Date: Tue, 18 Oct 2011 15:35:41 +0200 Subject: [asterisk-users] Problems during calls Hello dear list. We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when making calls, that the calls become silent.Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the conversation.When we then hangup, and redial immediately, the calls do not go through, we then have to try redial a couple of times, and then It suddenly gets through.There is nothing in the verbose log in Asterisk –r. SIP HW is Snom and Different types of Cisco. Anyone got an idea? Or at lest know how to dig deeper in logs? Med vennlig hilsen / Best regardsAbacus IT AS- din Visma Software Partner- your Visma Software Partner L.Aksel CelasunMobilnummer/cell phone: (+47) 900 15 103Sentralbord/Support 4000 1850ak...@abacus-it.no Se denne månedens gode tilbud fra Abacus IT AS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.7 and client outside network
One more thing can you post your peer's configs as you have it in the config file? and can you register with the same user from within the lan? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sun, 16 Oct 2011 12:33:27 +0200 From: ad...@tootai.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and client outside network Hi Tarek Le 15/10/2011 20:28, Tarek Sawah a écrit : Hello Daniel First question, do you have a firewall application or hardware installed on the network? The Asterisk server is also the firewall/router, iptables running on it. Second do you have some software similar to fail2ban? Yes, but I put the domain IP in ignoreip list. I checked fail2ban iptables rules, no trace of this IP Third check your IPTABLES if you can post the output of iptables-save would be good. if you can replace the localnet=Asterisk server external IP/32 with externip=Asterisk server external IP/32 I didn't send this info but externalip is setted to Asterisk server external IP/32 then we will be able to check your problem? This setup is working on tens of customers servers (1.2, 1.4 and 1.6), but this is the first one running 1.8 version. The same phone connect perfectly to our 1.6 server in the same conditions, so it's seems something related to 1.8 version. What I don't understand is that (violating IP ) should display the IP but in my case it's blank (or empty). Should domain contain as well the port despite the fact that we have insecure=port,invite? Thanks for your help Daniel Date: Sat, 15 Oct 2011 19:08:10 +0200 From: ad...@tootai.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and client outside network Hi, no clue on this? I found a thread in march from Faisal Hanif having the same problem but no one of the proposed ideas where working (reverse permit/deny, tried with only permit=0.0.0.0/0.0.0.0, aso), no luck :-) I don't now if it's solved for him. If someone had a solution on this, would be great to share ;-) Regards -- Daniel Le 07/10/2011 15:01, Administrator TOOTAI a écrit : Hi, my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:13597 parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by contact ACL (violating IP ) [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: Registration denied because of contact ACL doesn't matter if I connect through a VPN or to the public IP using STUN. My sip.conf: localnet=172.24.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Zero conf local network localnet=Asterisk server external IP/32 autodomain=yes ;allowexternaldomains=yes domain=172.24.30.250 ;Asterisk Server IP domain=Public Hostname domain=Another Public Hostname [309](snom320,ulaw-phone,callgroup1) type=friend insecure=port,invite secret=VoIP2auDIo contactdeny=0.0.0.0/0.0.0.0 contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as disallowed by contact ACL deny=0.0.0.0/0.0.0.0 permit=XX.XXX.XXX.XX/32 nat=yes Any clue? Why violating IP is empty? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.7 and client outside network
Hello Daniel First question, do you have a firewall application or hardware installed on the network? Second do you have some software similar to fail2ban? Third check your IPTABLES if you can post the output of iptables-save would be good. if you can replace the localnet=Asterisk server external IP/32 with externip=Asterisk server external IP/32 then we will be able to check your problem? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sat, 15 Oct 2011 19:08:10 +0200 From: ad...@tootai.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and client outside network Hi, no clue on this? I found a thread in march from Faisal Hanif having the same problem but no one of the proposed ideas where working (reverse permit/deny, tried with only permit=0.0.0.0/0.0.0.0, aso), no luck :-) I don't now if it's solved for him. If someone had a solution on this, would be great to share ;-) Regards -- Daniel Le 07/10/2011 15:01, Administrator TOOTAI a écrit : Hi, my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:13597 parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by contact ACL (violating IP ) [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: Registration denied because of contact ACL doesn't matter if I connect through a VPN or to the public IP using STUN. My sip.conf: localnet=172.24.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Zero conf local network localnet=Asterisk server external IP/32 autodomain=yes ;allowexternaldomains=yes domain=172.24.30.250 ;Asterisk Server IP domain=Public Hostname domain=Another Public Hostname [309](snom320,ulaw-phone,callgroup1) type=friend insecure=port,invite secret=VoIP2auDIo contactdeny=0.0.0.0/0.0.0.0 contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as disallowed by contact ACL deny=0.0.0.0/0.0.0.0 permit=XX.XXX.XXX.XX/32 nat=yes Any clue? Why violating IP is empty? Thanks for your help -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
i had a similar challenge having Asterisk listen to multiple ports.. some of my agents located in countries where SIP is blocked the only effective way is to use IPTABLES i believe your problem can be solved with the same method. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Wed, 12 Oct 2011 23:27:16 +0200 From: ge...@riseup.net To: asterisk-users@lists.digium.com Subject: [asterisk-users] Binding asterisk to two static IPs He all, I've got a similar setup like [1], and the same issues thats described there. However, there was never a reply to this thread. I'm using a HA-cluster to run asterisk, on two servers, with two virtual ips. One for the phones to register, the other one from a different net to send and receive calls trough my provider. This is aswell a private net, without nat. From [1]: If you put 0.0.0.0, it will bind to all addresses. In a HA Cluster, on the active node, if you have a box address of 192.168.1.101 and a floating address of 192.168.1.102, then if you use bindaddr=0.0.0.0 you will find that phones on the 192.168.1.x subnet will not register on the floating address, which of course defeats the point of HA clustering. What happens is that the registration packets go to the floating address 192.168.1.102 but the response packets appear to come from 192.168.1.101 [same NIC but the packet contains the base address attached to the NIC], so the registration fails. Any idea how to solve this? Thanks, Georg [1] http://www.fonality.com/trixbox/forums/trixbox-forums/help/binding-sip-multiple-not-all-ip-addresses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI not Installed?
what version of Asterisk are you using? try issuing agi show from the Asterisk CLI console and see if you get some output? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Wed, 12 Oct 2011 13:24:23 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] AGI not Installed? Hello Everyone, I am trying to get AGI going. The command agi show commands yields: DeadCommand Description No answer Not available Yes asyncagi break Not available No channel status Not available Yes database del Not available Yes database deltree Not available Yes database get Not available Yes database put Not available Yes exec Not available No get data Not available Yes get full variable Not available No get option Not available Yes get variable Not available No hangup Not available Yes noop Not available No receive char Not available No receive text Not available Norecord file Not available No say alpha Not available No say digits Not available No say number Not available No say phonetic Not available No say date Not available No say time Not available No say datetime Not available No send image Not available No send text Not available No set autohangup Not available No set callerid Not available Noset context Not available No set extension Not available No set music Not available No set priority Not available Yes set variable Not available Nostream file Not available Nocontrol stream file Not available No tdd mode Not available Yesverbose Not available No wait for digit Not available No speech create Not available No speech set Not available Yes speech destroy Not available Nospeech load grammar Not available Yes speech unload grammar Not available Nospeech activate grammar Not available No speech deactivate grammar Not available No speech recognize Not available No gosub Not available The /var/lib/asterisk/agi-bin/ dir is empty. Is there a ./config flag needed to install AGI? Thanks in Advance, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beep file with Record
i think you can try placing the beef file in the /var/lib/asterisk/sounds
directory and not the language specific one.
and your system is calling the beep file without having it in the dialplan?
sounds strange somehow to me.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
From: arjan.kr...@mobillion.nl
To: asterisk-users@lists.digium.com
Date: Wed, 5 Oct 2011 09:20:32 +0200
Subject: Re: [asterisk-users] Beep file with Record
CLI::-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full:
File beep does not exist in any format [Oct 4 16:19:38] WARNING[13370]:
file.c:950 ast_streamfile: Unable to open beep (format 0x8 (alaw)): No such
file or directory [Oct 4 16:19:38] WARNING[13370]: app_record.c:281
record_exec: ast_streamfile failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
I don’t call the beep file in my dialplan. Van:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:04
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record How are you calling the
beep.alaw from the dialplan?paste the relevant dialplan here and corresponding
CLI logs. On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:I placed a beep.alaw file in de directory, but
I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny
NicholasVerzonden: 04-10-2011 17:16Aan: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characters,
not xx/ (nl/fvdb) (feel free to correct my assumption that language has not
been expanded beyond the 2 character limitation)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten = s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
This is my complete CLI logging
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
0) in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
ast_openstream_full: File beep does not exist in any format [Oct 4
16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/service
line/${UNIQUEID) exten = s,n,Record(${A_serviceline_file}.wav,0,60)
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 16:30
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
Usually this message is received because you did something like
playback(beep.gsm) or playback(beep.wav) instead of playback(beep).
It is
(IMO) somewhat confusing because you have to do record(foo.gsm) but
you have to playback using playback
Re: [asterisk-users] Beep file with Record
can you post the while dialplan? it seems cropped somewhere as i dont' see it
starting or ending anywhere.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 5 Oct 2011 12:31:49 +0500
From: govoi...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Beep file with Record
hmmm...what i'm saying is this
exten = s,n,Set(CHANNEL(language)=en))
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)exten
= s,n,Record(${A_serviceline_file}.wav,0,60)
exten = s,n,Set(CHANNEL(language)=nl))
On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
Yes I already try this (only with language nl)
exten = s,n,Set(CHANNEL(language)=nl))
I also try to place the voicefile in the directory /var/lib/asterisk/sounds/
and /var/lib/asterisk/sounds/applications/ of but without any success.
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:26
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
Since you've changed the language (sound directory) So just as a test change
the language back to en and if it goes well revert back language after the
recording.
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
CLI::-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format [Oct 4 16:19:38] WARNING[13370]: file.c:950
ast_streamfile: Unable to open beep (format 0x8 (alaw)): No such file or
directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)I don’t call the beep file
in my dialplan.
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:04
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?paste the relevant
dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 17:16Aan: 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characters,
not xx/ (nl/fvdb) (feel free to correct my assumption that language has not
been expanded beyond the 2 character limitation)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten = s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
This is my complete CLI logging
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
0) in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
ast_openstream_full: File beep does not exist in any format [Oct 4
16
Re: [asterisk-users] USA Did required
Google is your best friend when looking for this type of assistance my friend. try callcentric vonage packet8 for reliable retail DIDs. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sat, 1 Oct 2011 00:51:59 +0530 From: amit.magn...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] USA Did required Hello members, I am looking for USA incoming DID which can be registered on softphone/IP Phone/ Pap2 devices. The DID will only be required to receive inbound calls and no outbound calls. Let me know your best per month prices/cost for the above. Regards, Amit Mehta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Bull Service Providers
for some reason i don't think (unlimited incoming channels) fits with (dirt cheap DIDs) as you will be abusing their network .. they should start charging per minute .. or you should pay for extra channels several DID providers would offer you 20 channels per did at some rate of 9$ a month per did.. 5 Euros per month and you should pay Extra for Extra channels.. could be the same amount for the same amount of channels Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 11:09:10 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Bull Service Providers
What does (international long) mean exactly? are you a calling cards company? if so you should look for some company that will be charging you like 0.004 Cents per minute.. and you can find companies that will add more channels to your DID. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 11:15:13 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers I should have mentioned we are interested in international long distance. That will be a big part of our business. Cheers, Nick. On Thu, Sep 29, 2011 at 11:12 AM, Danny Nicholas da...@debsinc.com wrote: They aren't everywhere, but we have had good experience with Voicepulse and their rate is typically less than $0.015 per minute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
Re: [asterisk-users] No Bull Service Providers
I have no knowledge of any commercial brand that operates in that region and would offer DIDs in those countries.. AND your channel requirements are a bit limited by technology in those regions... and VoIP termination Legislation in those countries whether they allow Calling Cards business, allow DID sales. those issues have more effect on your business. could have helped in US DIDs.. but in Asia i'm no aware of the presence of such providers. however TATACOMMUNICATIONS is the largest VoIP Operating entity in that region and you may find some luck contacting them? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 11:24:43 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers Hello Tarek, For channels, usually they charge per additional channels. I guess being more explicit what it comes down to is: * Reliable service * Agressive Pricing * For DIDs - International Coverage - Per Aditional Channel Pricing * For SIP Termination - International Rates - Per additional trunk pricing We are looking to provide large scale long distance service to thrid world countries such as Sri Lanka, Philippines, India, Pakistan etc... So would require DID for those reagions with the channel support, and sip termintation to Canada and the US with trunk support. Nick. On Thu, Sep 29, 2011 at 11:13 AM, Tarek Sawah tareksa...@hotmail.com wrote: for some reason i don't think (unlimited incoming channels) fits with (dirt cheap DIDs) as you will be abusing their network .. they should start charging per minute .. or you should pay for extra channels several DID providers would offer you 20 channels per did at some rate of 9$ a month per did.. 5 Euros per month and you should pay Extra for Extra channels.. could be the same amount for the same amount of channels Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 11:09:10 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers Very true... But there should be an equilibrium, the relaiable service, and aggressive pricing comes to meet? Guys please share your experiences. Cheers, Nick On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich c.savinov...@itntelecom.com wrote: In my professional opinion, the phrases I don't want no Bull service and I want the cheapest service are total contradictions. Down the road something is not going to give. C. Savinovich On September 29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com wrote: This belongs on the commercial list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, September 29, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Bull Service Providers Hello Everyone, We are looking for DID and SIP Termination service providers. Since there are so many these days, can you guy mention the BIG players that are supplying the rest of the little guy? We are looking for the cheapest, and scaleable infrastructure (i.e. unlimited channels for DID, and trunks for termintation). To summarize we are looking for the major players in the DID and SIP Trunk market, no/limited headache. This is for wholesaler service. Thanks in Advance, Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] No Bull Service Providers
one thing i'm sure of? Honesty is a waste in this type of business.. all the features youa re talking about .. have been offered and tested with customers.. the bottom like .. when a customer buys a 2$ calling card . he expects to make a call and say his words and hangs up .. all those features won't be of use for him for a card that will allow him to talk as much minutes as he can! you abusing free routes or not.. is not his business actually. those features can be offered to PINLESS customers who can pay 100-300 $ per account! Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Thu, 29 Sep 2011 12:03:26 -0400 From: sym...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Bull Service Providers You will notice on the calling card shelves there are only a handful of companies producing lots of different cards. I have! That's what led me to CC for starters, then implementing a more novel startup product. But. Regardless of all the corruption, my goal is to offer something honest TRUTH, I like that ;), reliable and as consistent as possible. We cannot compete against free, but we can try our best. Again, CC is just an entry point, we can doing this like: speech to text - Natural Language Processing (NLP) - text to speech. Bringing computer science to VoIP. This is our long term.. I just need to keep the investor happy for now.. Nick On Thu, Sep 29, 2011 at 11:48 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 29 Sep 2011, Nick Khamis wrote: Hello Jeff, There will always be fierce competition, we are starting of with prepaid for an obvious source of quick revenue, we will also be rolling out a few more products in the next year.. It seems like they LIE about their LD rates. A company in Australia was charged with this not too long ago. Stolen minutes? Not that I would be interested in stealing! I just want to be educated in such an act. You will often see discussion on this list about asterisk servers being compromised and the result being very expensive calls placed until the compromise is noticed and shutdown. Those calls are placed by nefarious wholesalers that take advantage of the free routes they manage to find as long as possible. Hard to compete against free! Other games the calling card companies play - they will release a card with unbelievable rates so that it quickly gains market share, then slowly back off the minutes offered by the card (without changing the rate sheets of course) until it is noticed by the consumers, who stop buying it. Then that card is discontinued and another is produced in the same manner. You will notice on the calling card shelves there are only a handful of companies producing lots of different cards. There are many more tricks they use to dupe the consumers and stifle competition. Hidden or non-disclosed connection rates, maintenance fees charged every few days to burn off credit on the cards, time restrictions on the lower rates, etc. We actually produced a card once we called TRUTH (which was honest about rates, had no hidden fees, etc) and it sold ok for a while, but when the card next to it on the shelf claims twice the minutes for the same $$$, eventually they win. In the end this business doesn't make money unless you are selling millions of minutes per month, and even then the margins are slim and you have to play the same games to compete. What we thought would be a fairly easy business to run became a maintenance nightmare, and a single instance of fraud could wipe out months worth of profits. A2billing didn't exist when we started, so we rolled our own. Seems pretty popular now - maybe it would work well for you. Good luck, j Of course we don't have to use the same in/outbound providers. I should have been clearer about that. You mentioned Least Cost Route/Rate (LCR), any reason why you did not use what is already out there? Provided by a2billing etc...? We can also implement something using AGI if needed Nick. On Thu, Sep 29, 2011 at 11:21 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 29 Sep 2011, Nick Khamis wrote: I should have mentioned we are interested in international long distance. That will be a big part of our business. It sounds like you are intending to start a calling card company. Good luck - the competition is fierce, and you will be competing against companies that outright lie about the capacity of their cards, and use stolen minutes to fulfill them as often as they can. If you intend to do wholesale by reselling, you don't need to use the same company for inbound and outbound. In fact for outbound you will probably have many upstream providers, as your goal will be to find the cheapest reliable route
Re: [asterisk-users] Receiving musinc on hold instead of ring
this is related to your carrier's SIP messages as they are sending a sendonly attribute instead of sendrecv (taking a wild guess here) your asterisk will act as if the call was placed on hold thus the MOH butts in. an sip debug log for a similar call will be more helpful? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: alexreca...@gmail.com Date: Wed, 28 Sep 2011 03:44:35 +0200 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Receiving musinc on hold instead of ring Hi all and thanks for reading. I am having a very strange issue. When dialing out with a certain carrier, asterisk 1.6.20 will play music on hold instead of a ring tone, although this behaviour is NOT what I want. Example dialplan execution: -- Executing [0021266xxx@test:13] Progress(SIP/100-1e04, ) in new stack -- Executing [0021266xxx@test:14] Dial(SIP/100-1e04,SIP/21266xxx@x.x.x.x) in new stack -- Called 21266xxx@x.x.x.x -- Call on SIP/x.x.x.x-1e05 placed on hold -- Started music on hold, class 'default', on SIP/100-1e04 -- SIP/x.x.x.x-1e05 is making progress passing it to SIP/100-1e04 Now, a SIP packet capture shows no trace of the call being put on hold! Sample wireshark capture for the same call: x.x.x.x - y.y.y.y SIP/SDP Request: INVITE sip:21266xxx@x.x.x.x, with session description y.y.y.y - x.x.x.x SIP Status: 100 Giving a try y.y.y.y - x.x.x.x SIP/SDP Status: 180 Ringing, with session description And I get the music on hold instead of the ringtone. I have tried placing Progress() in front of Dial() but to no avail. I do not want to use the r option in Dial() because then I lose the destination ringtone in early media which is important to my customers. Anybody had a similar issue? Any idea of what parameters I can try to tweak, as I am stumped... Thanks! Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receiving musinc on hold instead of ring
i have faced this problem with one of the major VoIP whole providers in India .. they have a new platform with Sonus switches.. which does not support sendrecv media attribute .. however a work around that may work for you .. is enabling re-invite on their peer. let me know if this works out for you. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: alexreca...@gmail.com Date: Wed, 28 Sep 2011 18:59:39 +0200 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Receiving musinc on hold instead of ring this is related to your carrier's SIP messages as they are sending a sendonly attribute instead of sendrecv (taking a wild guess here) your asterisk will act as if the call was placed on hold thus the MOH butts in. an sip debug log for a similar call will be more helpful? Thanks for the answer Tarek! I will try to obtain a full SIP trace tonight. If the problem is indeed that the carrier is sending the sendonly attribute in the SDP instead of sendrecv, what can I do? Is there anything I can configure on my side? Thanks again, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit outbond calls duration to 1 minute
have a look at the following:
L(x[:y][:z]): Limit the call to 'x'
ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is
required, 'y' and 'z' are optional.
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 17:59:27 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
hello list
i have configured a sip account in order to do an outbound calls and i want to
force a hang up after 1 min for 222 sip
in extensions.conf i have
exten = 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 222,n,AbsoluteTimeout(60)
exten = 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 222,n,Dial(SIP/${EXTEN},,KkTt)
exten = 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
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Re: [asterisk-users] Limit outbond calls duration to 1 minute
exten = 222,n,Dial(SIP/${EXTEN},,KkTtLL(6:3:1))
this will call the extension and sets the limit to 6MS which equals 60
seconds.. and will inform the caller of his remaining time when he has only 30
seconds left.. and will repeat the notification every ten seconds (this is an
over do and playing such sounds files at this rate will consume the resources!)
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 18:22:57 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
but there is no exemple for when i must put X in order to limit the call
can you please give me an exemple
regards
2011/9/28 Tarek Sawah tareksa...@hotmail.com
have a look at the following:
L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated
every 'z' ms) Only 'x' is required, 'y' and 'z' are optional.
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 17:59:27 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
hello list
i have configured a sip account in order to do an outbound calls and i want to
force a hang up after 1 min for 222 sip
in extensions.conf i have
exten = 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 222,n,AbsoluteTimeout(60)
exten = 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 222,n,Dial(SIP/${EXTEN},,KkTt)
exten = 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
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Re: [asterisk-users] Limit outbond calls duration to 1 minute
one adjustment i would suggest is using (|) instead of (,)
exten = 222,n,Dial(SIP/${EXTEN}||KkTtL(6))
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 18:32:28 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
sorry but the issue still the same there is no hangup after 1Min
regards
2011/9/28 Danny Nicholas da...@debsinc.com
As I read this, the following should be correct:
exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6))
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Wednesday, September 28, 2011 1:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
but there is no exemple for when i must put X in order to limit the call
can you please give me an exemple
regards
2011/9/28 Tarek Sawah tareksa...@hotmail.com
have a look at the following:
L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated
every 'z' ms) Only 'x' is required, 'y' and 'z' are optional.
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 17:59:27 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
hello list
i have configured a sip account in order to do an outbound calls and i want to
force a hang up after 1 min for 222 sip
in extensions.conf i have
exten = 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 222,n,AbsoluteTimeout(60)
exten = 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 222,n,Dial(SIP/${EXTEN},,KkTt)
exten = 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
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Re: [asterisk-users] Queuing: calls stay in queue and agents are ready !!
Bilal , if you can do a core show queue QUEUENUMBER and paste the output here at the moment of this problem it will be helpful to see what is the status of your agents at that moment for the queue. does it help if the agents logout then back in instead of disconnecting the call and calling back again? what is the timeout for the agent setup in the queue settings? or more helpful if you paste your queue settings Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Mon, 19 Sep 2011 02:46:59 -0700 From: bilmar...@yahoo.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queuing: calls stay in queue and agents are ready !! Hi All; I configured some queues, and I configured the dialed numbers for login and logout for the agents. Two agents are logged in, the first two calls are received at the agents and they answered and hangup. Again, the two agents are idle and ready to receive calls. The third and call goes to queue and stay in waiting although the agents are ready !!! We disconnect the call and re call again, the same thing (the call goes for the queue in the waiting and does not go for the agents who are ready to receive calls). I tried to change something in the settings, I made the autofill=yes and the autopause=no without any success (the same problem). What could cause for such behaviour? What is the parameter or the settings that make such thing happen? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] redundant traffic
I would suggest using a Vyatta based server to Run Asterisk on or behind.. and use the load balance feature to forward your incoming connections to the Asterisk server this will create one default gateway for your asterisk server so you won't have to have two separate networks identified.. nor two NICs. or identify two ports on the server forwarding one of them to the original binding port of Asterisk. if it wasn't for the Default gateway .. it would have been easy to do some port forwarding on the internet router side. but Asterisk needs to communicate with the internet to send packets back. this is one of the scenarios i can think of. and can be done in 20 minutes. well it can be expensive if you calculate the costs of an additional computer on the network. :S Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: chayn...@gmail.com To: asterisk-users@lists.digium.com Date: Sat, 17 Sep 2011 17:31:56 -0400 Subject: [asterisk-users] redundant traffic Hello, I’ve got a customer that wants me to set up their single Asterisk server so that they can receive redundant traffic streams from their origination provider. They want the traffic broadcast to 2 static IP addresses on the Asterisk server for redundancy. Their they want to be sure to receive traffic if one of their subnets/gateways goes down. As I understand it, having the two IP's set up to receive redundant information as possible in Linux, but I wonder how (or if it's even possible) to address this in Asterisk. As anybody ever done this? Claude -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reporting for Asterisk Call Center
actually Bilal, the Asteirsk CDR reports are placed on a different Database than the configurations .. you will need to install asterisk-addons which includes a module for cdr reporting to MYSQL DB. so you don't have to do the configs from the DB at all second.. in regards to the Flash Operator Panel you can have a look at a demo here: http://www.asternic.org/demo.php its a nice web interface gives you a live look at your call center .. who is active who is idle .. how many in Queue .. who is online and who is offline.. what trunks are busy ... etc third: theoretically you can set all Asterisk boxes to load from one database server (never done it myself).. actually it's one of the methods used for redundancy (somebody correct me if i'm wrong?). my only concern is with DB Management systems there is what we call it LOCK where a process locks the whole db or a part or it in order to do it's manipulation.. so i'm not sure if the database will be locked by one of the asterisk boxes when writing to it? which prevents the rest from writing to it at the same time? regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Tue, 13 Sep 2011 02:43:05 -0700 From: bilmar...@yahoo.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reporting for Asterisk Call Center Dear Tareq; I am not using mysql, the configuration on the text configuratoin files and the logs are existed under the directory (/var/log/asterisk). Well, to use mysql: then it means the configuration will be also in the database or I can use mysql only for reporting? What is the Flash Operator? By the way, I have another question if you can help me if you used the database with sql, actually I was facing one time a case and maybe the Database usage will help me if you can advise me: If I have multiple Asterisk servers are running, and I need them to work centralized (I mean from one configuration) so to work as one system, then if I have database for configuration, I can acheive this by making all the servers read and write the configuration from the database server? Thanks for your help Tareq. Regards Bilal --- those reports can be easiely extracted from the MYSQL database my friend.. and you can add the Flash Operator Panel if you want to monitor live activities like how many in queue and how many ON CALL .. etc anyway the Elastix is a stand alone distribution you can find more info and downloads at : http://elastix.org/ regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 not accepting call from DID
you didn't provide your dialplan for the incoming call context from_poland? nor registration string? could be a dial plan problem .. or codec issue.. as long as you register properly the server has no problem with NAT.. it's a routing or codec issue i think. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Mon, 5 Sep 2011 19:50:34 -0600 From: syscon...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 not accepting call from DID It seems to me nat=yes is not working correctly in asterisk 1.8.5 rtp set debug on shows: Got RTP packet from 10.0.0.110:6000 (type 00, seq 029667, ts 2129095321, len 000160) Sent RTP packet to 10.0.0.110:6010 (type 00, seq 065112, ts 2129095320, len 000160) I've tried 'nat=yes' 'nat=comedia' it makes no differece. -- Joseph On 09/05/11 15:00, Joseph wrote: I have DID, it registers OK with the provider, but when I try to call this number (it suppose to ring my Asterisk) asterisk 1.8 does not respond. sip show peers Name/username Host Dyn Forcerport ACL Port Status actio-out/48746612254 81.15.150.20 N 5060 OK (201ms) sip.conf part: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 useragent = Centrala [actio-out] type=friend secret= user=48746612254 username=48746612254 fromuser=48746612254 authname=48746612254 callerpage=48746612254 fromdomain=sip.actio.pl host=sip.actio.pl insecure=port,invite nat=yes qualify=yes dtmfmode=inband disallow=all allow=ulaw allow=alaw context=from_poland canreinvite=no The setting above worked OK with Asteriks 1.4 Here is debug info, which I don't know how to interpret. -- Executing [901148746612254@internal:1] Dial(SIP/11-0002, SIP/901148746612254@pstn-1270,60,tr) in new stack [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:25695 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) == Using UDPTL CoS mark 5 [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:7496 sip_alloc: Allocating new SIP dialog for 5a2cdf8339e0ad2911ad393036c05165@127.0.0.1:0 - INVITE (No RTP) [Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:347 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x88c3b10' [Sep 5 14:04:35] DEBUG[26209]: res_rtp_asterisk.c:474 ast_rtp_new: Allocated port 16690 for RTP instance '0x88c3b10' [Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:356 ast_rtp_instance_new: RTP instance '0x88c3b10' is setup and ready to go [Sep 5 14:04:35] DEBUG[26209]: res_rtp_asterisk.c:2372 ast_rtp_prop_set: Setup RTCP on RTP instance '0x88c3b10' == Using SIP RTP CoS mark 5 [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:4928 do_setnat: Setting NAT on RTP to Off [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:4936 do_setnat: Setting NAT on UDPTL to Off [Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:1459 ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 'SIP/pstn-1270-0003' with that of 'SIP/11-0002' [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable SIPURI. [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:5463 sip_call: Outgoing Call for 901148746612254 [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:10989 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:10990 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:3054 initialize_initreq: Initializing initreq for method INVITE - callid 770d283f78ef7d00782d2dd043212ed2@10.0.0.103:5060 -- Called SIP/901148746612254@pstn-1270 [Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '770d283f78ef7d00782d2dd043212ed2@10.0.0.103:5060' Request 102: Found [Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '770d283f78ef7d00782d2dd043212ed2@10.0.0.103:5060' Request 102: Found [Sep 5 14:04:35] DEBUG[26083]: rtp_engine.c:538
Re: [asterisk-users] Anyone using Asterisk on VirtualBox ?
i did do some Asterisk tests on SUN VBOX .. works like a charm but you need to dedicate some good resources to the virtual box! Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: zhulizh...@live.com To: asterisk-users@lists.digium.com Date: Fri, 2 Sep 2011 08:37:55 + Subject: Re: [asterisk-users] Anyone using Asterisk on VirtualBox ? hi: please check the redfone solution. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: aster...@a-domani.nl To: asterisk-users@lists.digium.com Date: Thu, 1 Sep 2011 23:48:46 +0200 Subject: Re: [asterisk-users] Anyone using Asterisk on VirtualBox ? On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote: My main interest of being on Virtual platform is portability / Backup. In case of any h/w issues, or crashes, simply copy the VM on to another box and you are up in minutes. Sanjay -- Doing that right now, although in my case i use XEN. Besides being hw independant, it is easier to play with a different version for a while (1.4 / 1.6.0 / 1.6.1 / 1.6.2 / 1.8.0) and being able to switch back in minutes. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
try to look for N82 nokia mobile devices.. you get the benefits of a Mobile device with it's phone book and mobility features (games when you are bored :P) .. and other features.. and the native SIP client works fluently with no problems at all supporting almost commercial codecs like (G729).. and it works with WIFI.. i use it at home. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sat, 27 Aug 2011 10:14:24 +0100 From: gordon+aster...@drogon.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ? On Sat, 27 Aug 2011, Alan Lord (News) wrote: On 26/08/11 19:02, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. We've been using the Siemens Gigaset 685IP range for over three years and I'm (still) very pleased with them: +1 The base station is separate from the handsets - which is typically different from most DECT setups - the plus point is that you can position the base in a good location - ie. high on a wall, rather than anywhere else. Another plus is that the base has a single built-in ATA, so it can connect to the home PSTN line. The base also has an Ethernet socket to connect to the LAN and it can have up to 6 SIP accounts - each handset (up to 6) can be configured to ring on a particular SIP account or many SIP accounts and/or the PSTN line. Each handset has a default SIP account (or PSTN) to make outgoing calls on, but you can select any other SIP account or the PSTN by appending a code to the number you dial. They are very flexible - and being DECT, have superb range. I've installed many of these for my customers - typically the home office types - where they only want one phone on their desk - so the same handset can answer their home phone or their office SIP account, while providing wireless handsets throughout the rest of the house. A limitation is that one base can only handle 2 simultaneous SIP calls (plus a call via the PSTN), so if 2 phones are in-use, then the system can't take a 3rd call, however that's rarely a limitation in a domestic environment. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8
Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8
Actually i had to upgrade to 1.6 due to a provider problem with session-timers and RTP data .. then i downgraded again to 1.4. do you suggest that i test 1.8 instead of 1.6? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 12 Sep 2011 10:54:35 -0500 Subject: Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 I personally would not bother with 1.6 unless you needed some feature in that branch. 1.4 is the stable branch, but it seems that all of the resources are being channeled into 1.8 and 10.0, so 1.6 is a rabbit hole you really shouldn't be headed into. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 12, 2011 10:19 AM To: Asterisk Users Subject: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reporting for Asterisk Call Center
if you provide what kind of reporting you need it would be easier to point a few pointers? either you can build it yourself.. or try the Call Center module from Elastix.. can be a good tool Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sat, 10 Sep 2011 10:28:00 +0300 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reporting for Asterisk Call Center On Fri, Sep 09, 2011 at 01:28:28PM -0500, Gerardo Barajas wrote: There are a lot of reporting tools. I have used: Asternic: http://www.asternic.biz/ QueueMetrics: http://queuemetrics.com/index.jsp Non of those are Free (Open Source). -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] out of the blue one way audio
Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet link of 1Mbps Dedicated Leased Line. 3- Cisco Router 4- Hosted Asterisk server (Asterisk 1.4.40.1 x64 bit 8 GB ram, Intel(R) Xeon(R) X3210 @ 2.13GHz CPU) 5- additional SIP Soft phones in several locations over the world (Zoiper, X-Lite, Nokia Native Sip). 6- Packet8 Sip trunking for Inbound calls 7- IDT (Net2Phone) Sip Trunk for outbound calls. (two IPs) Network Profile: Cisco Router has a Public IP of 196.XXX.XXX.XXX and a private IP 192.168.100.245 computers have IP addresses : 192.168.100.XXX/24 default gateway: 192.168.100.245 DC: 192.168.100.2 DNS: 192.168.100.2 PROXY Server: 192.168.100.2 (Forced in Internet Explorer) Voip Traffic going directly from 192.168.100.245 Http Traffic goes to 192.168.100.2 then via another internet link (ADSL 8bps connection) Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0.0/0.0.0.0 type=friend secret=PASSWORD qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=gsm context=from-callcenter canreinvite=no we have a call recording for outbound and inbound calls. the problem is not happening on all calls at once.. it happens on random extensions at random times and random durations however most noticeable durations are around 7 minutes and 20 minutes (most occurring) one additional situation.. the original bind_port for asterisk server is 5060 however after three or four hours of operating on that port the computers unregister and are unable to make calls at all .. or even register we changed the port to 5070 and things are working properly now. although this port issue is only noticeable on the above setup and on that facility only. other internet links are able to provide stable connection over 5060. any additional information can be provided. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] out of the blue one way audio
this is happening on all Soft phones are facing the same problem. Zoiper , X=lite , our own pjsip based dialer (CRM). this was not the issue .. it happened suddenly .. we switched internet links even. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Mon, 2 May 2011 14:45:58 +0300 From: hatemm...@gmail.com To: asterisk-users@lists.digium.com CC: yamennaj...@ids-tech.net Subject: Re: [asterisk-users] out of the blue one way audio Check if this problem happening with xlite useres only i remember there is option in xlite causing this problem On May 2, 2011 2:36 PM, Tarek Sawah wrote: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] out of the blue one way audio
because they are behind a router and using private IP addresses. and the Cisco router is Nating our traffic Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 2 May 2011 08:11:23 -0400 Subject: Re: [asterisk-users] out of the blue one way audio Why nat=yes ? -- Sent from my iPhone On May 2, 2011, at 7:33 AM, Tarek Sawah wrote: Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet link of 1Mbps Dedicated Leased Line. 3- Cisco Router 4- Hosted Asterisk server (Asterisk 1.4.40.1 x64 bit 8 GB ram, Intel (R) Xeon(R) X3210 @ 2.13GHz CPU) 5- additional SIP Soft phones in several locations over the world (Zoiper, X-Lite, Nokia Native Sip). 6- Packet8 Sip trunking for Inbound calls 7- IDT (Net2Phone) Sip Trunk for outbound calls. (two IPs) Network Profile: Cisco Router has a Public IP of 196.XXX.XXX.XXX and a private IP 192.168.100.245 computers have IP addresses : 192.168.100.XXX/24 default gateway: 192.168.100.245 DC: 192.168.100.2 DNS: 192.168.100.2 PROXY Server: 192.168.100.2 (Forced in Internet Explorer) Voip Traffic going directly from 192.168.100.245 Http Traffic goes to 192.168.100.2 then via another internet link (ADSL 8bps connection) Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0.0/0.0.0.0 type=friend secret=PASSWORD qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=gsm context=from-callcenter canreinvite=no we have a call recording for outbound and inbound calls. the problem is not happening on all calls at once.. it happens on random extensions at random times and random durations however most noticeable durations are around 7 minutes and 20 minutes (most occurring) one additional situation.. the original bind_port for asterisk server is 5060 however after three or four hours of operating on that port the computers unregister and are unable to make calls at all .. or even register we changed the port to 5070 and things are working properly now. although this port issue is only noticeable on the above setup and on that facility only. other internet links are able to provide stable connection over 5060. any additional information can be provided. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP SHOW REGISTRY SHOWS NOTHING
Greetings i've setup a new asterisk server 1.4.38 ... everything works fine however i need to register the server with another SIP provider.. the registration string .. the server is not attempting to register .. sip show registry shows nothing.. i created an sip_registration.conf file and asterisk is parsing it.. but nothing shows in the sip show registry.. any one can say why? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Billing Question???
If you look at it the way you want it.. you usually tell your customer the
available funds and minutes in their account right?
How will you explain politely that you have dropped their calls for lack
of balance because someone else used their account?
If you don't tell them their balance and call duration before call .. then
that won't be a problem.
Now you can do some kind of script to do the math and disconnect calls when
balance is over.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, October 21, 2010 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Realtime Billing Question???
Hi Sherwood ,
well , i think you did not understand my question , i want real time billing
like as i mentioned that if i want to dial 5 number with different call rate
how can i access same
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested it will assign total balance to those 5
people but actually we can not do like this as total balance of user $100 ,
as per your suggestion it will give $100 for those 5 people which is
practically wrong i think.
give your thougts.
regards
dhaval
On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello All,
after so long time i posted a new question regarding billing, hope anyone
have some solution.
I have situation in that i want to do billing of more than 1 call in real
time below are scenario and explanation.
Scenario:
A customer called my DID number and after that from here i dial few number
let say 5 number. once number are placed into DIAL
i will put this customer into conference [MEETME] , once a Members are
picked up call they will also patched into conference and
talking is started, every thing working fine with DIAL-PLAN and DB look up.
Now, i want to do billing on customer dialed my DID, and from that actually
it DIALED 5 numbers, how can i DO real time billing
into this situation, like numbers can be different It can be ISD,STD,Local
and also free .
if customer having initial balance of $100 then how can i check balance
every time.in a situation once balance is nil then i want to disconnect
calls . is any one facing this type of situation.
give me some idea ,
regards
Dhaval
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Dhaval,
This sounds very much like a system I'm working on for a client right now.
I'm not permitted to disclose much about it due to the NDA i signed, but
I'll risk giving you a point in the right direction.
First, you should create a table in your database that has a column called
callid, and other columns that you will have to decide upon. This table will
be called something like 'call_references'. Oh, and you'll want to define
callid as the primary key for records in that table, but DO NOT make it an
autoincrement, you're going to populate it with a value that is described in
the next step.
Second, at the beginning of the original call you mentioned, define a
variable that will be unique to that call. I personally have done this by
stripping all non-digits from the caller's callerid (using
Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
Next (this is where I have to start being a bit vague), you're going to
perform an INSERT query, creating a new call_references record (using that
variable I just showed you how to construct as callid's value).
Now, when you defined that variable, you should have preceded the variable
name with two underscores ( __ ), which will tell Asterisk that channels
spawned by the current channel will inherit that variable and it's value.
Voila, you now have a method for storing realtime data such as billing
information between MULTIPLE calls.
I wish I could tell you more, but I can't violate my client's Non-Disclosure
Agreement.
Hope this helps you out!
Sherwood McGowan
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Re: [asterisk-users] Asterisk Realtime Billing Question???
actually my mail was not meant to be disrespectful. it was an inquiry. i have a billing system and had a few of those thoughts regarding real time billing. my issue was explaining to a customer that his call disconnected an hour earlier because someone else used his account.. I'm doing retail not wholesale, you may understand my question more clearly now? Regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: sherwood.mcgo...@gmail.com Date: Thu, 21 Oct 2010 05:18:17 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Realtime Billing Question??? Tarek, I'm not sure why it would be our problem is someone came into your office and started making long distance calls over a trunk I was providing your company I'm pretty sure that if I had tried that with some of my carriers in the past they would have laughed until they cried... Oh, and also, since this was a wholesale carrier, the customers were in control of their own freeze amount. It was there to allow THEM to control their account better. I'd be willing to bet that my clients would have been happy to just keep billing them for every minute they used. Lastly, I would like to just say, I'm not the guy who requested the feature, I'm the guy who figured out how to make it happen, and making it happen back in early 2006, when the MySQL addon was just BARELY stable... It's ok, I don't need respect, I have the knowledge that I'm the mick, and I'm awesome :P Cheers :D On Thu, Oct 21, 2010 at 4:37 AM, Tarek Sawah wrote: If you look at it the way you want it.. you usually tell your customer the available funds and minutes in their account right? How will you explain politely that you have dropped their calls for lack of balance because someone else used their account? If you don't tell them their balance and call duration before call .. then that won't be a problem. Now you can do some kind of script to do the math and disconnect calls when balance is over. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, October 21, 2010 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Realtime Billing Question??? Hi Sherwood , well , i think you did not understand my question , i want real time billing like as i mentioned that if i want to dial 5 number with different call rate how can i access same balance into those 5 people, if all are connected how can i periodically update billing , as you suggested it will assign total balance to those 5 people but actually we can not do like this as total balance of user $100 , as per your suggestion it will give $100 for those 5 people which is practically wrong i think. give your thougts. regards dhaval On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan wrote: On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA wrote: Hello All, after so long time i posted a new question regarding billing, hope anyone have some solution. I have situation in that i want to do billing of more than 1 call in real time below are scenario and explanation. Scenario: A customer called my DID number and after that from here i dial few number let say 5 number. once number are placed into DIAL i will put this customer into conference [MEETME] , once a Members are picked up call they will also patched into conference and talking is started, every thing working fine with DIAL-PLAN and DB look up. Now, i want to do billing on customer dialed my DID, and from that actually it DIALED 5 numbers, how can i DO real time billing into this situation, like numbers can be different It can be ISD,STD,Local and also free . if customer having initial balance of $100 then how can i check balance every time.in a situation once balance is nil then i want to disconnect calls . is any one facing this type of situation. give me some idea , regards Dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dhaval, This sounds very much like a system I'm working on for a client right now. I'm not permitted to disclose much about it due to the NDA i signed, but I'll risk giving you a point in the right direction. First, you should create a table in your database that has a column called callid, and other columns that you will have to decide upon
Re: [asterisk-users] Asterisk Redundancy
Has any of you tested Vyatta Load balancing and fail over solution with Asterisk? It uses heartbeat and works like magic with regular traffic but didn't have the time nor chance to test it with VoIP traffic.. but I think it's the same way. Anyone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen Sent: Monday, September 27, 2010 5:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Redundancy Michelle Dupuis mdup...@ocg.ca writes: Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com (also on the voip wiki) You get the cluster/heartbeat replication without needing to add openSER or full HAlinux. A simpler approach - easier to config and manage How do you handle replicating voice mails? /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to pick a codec on the fly
i think it's SIP_CODEC now .. and not _SIP_CODEC? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: da...@debsinc.com To: dan...@tryba.nl; asterisk-users@lists.digium.com Date: Mon, 27 Sep 2010 13:30:08 -0500 Subject: Re: [asterisk-users] How to pick a codec on the fly -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba Sent: Monday, September 27, 2010 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to pick a codec on the fly On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote: I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI-- IAX testing. exten = 1234,1,Set(_SIP_CODEC=alaw) exten = 1234,n,Goto(0234,1) exten = 2234,1,Set(_SIP_CODEC=slin) exten = 2234,n,Goto(0234,1) Should do the trick. -- Daniel Tryba Nice try, Daniel, but apparently _SIP_CODEC is no longer useful in 1.4X. -- Executing [...@from-pstn:7] Goto(DAHDI/1-1, default|s|1) in new stack -- Goto (default,s,1) -- Executing [...@default:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@default:2] Goto(DAHDI/1-1, select-func|s|1) in new stack -- Goto (select-func,s,1) -- Executing [...@select-func:1] WaitExten(DAHDI/1-1, 5|m) in new stack -- Started music on hold, class 'default', on DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 == CDR updated on DAHDI/1-1 -- Executing [...@select-func:1] Set(DAHDI/1-1, _SIP_CODEC=ulaw) in new stack -- Executing [...@select-func:2] Dial(DAHDI/1-1, IAX2/xxx/332|30|m) in new stack -- Called xxx/332 -- Started music on hold, class 'default', on DAHDI/1-1 -- Call accepted by XXX.XXX.XX.XX (format gsm) -- Format for call is gsm -- IAX2/ffb-18075 answered DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1 -- Hungup 'IAX2/xxx-18075' == Spawn extension (select-func, 2, 2) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ODBC Insert issue
DID you grant your user the ability to INSERT into the MSSQL db?
I have asterisk inserting easily
Just a privileges issue
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand
Sent: Sunday, September 26, 2010 9:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk ODBC Insert issue
Hi guys,
Having issues with doing an insert statement using ast 1.4.24:
[START]
dsn=mssql-asterisk
write=INSERT INTO testdb (callarrival,callerid) VALUES
('${VAL1}','${VAL2}')
SET(ODBC_START()${TIMESTAMP},${CALLERID(num)})
No errors pop up on execute, but nothing gets inserted.
Read and update work fine,
Wondering where I'm going off track with this,
Thanks,
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Re: [asterisk-users] differential billing
if you are deploying your own system.. then you can build a small application (AGI) that would do the math for you .. will devide the call duration into the stages you want .. and does the calculation.. i think MYSQL already can do that.. but a PHP script will do it faster and easier.. or like our billing system.. C# application interacting with Asterisk doing all the math. after all it's all SQL and Asterisk working. you can do that with a dial plan i believe.. so why not build an AGI to do it for you? -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +13864929993 From: basit.e...@gmail.com Date: Sat, 25 Sep 2010 23:27:56 +0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] differential billing Tarek, I already tested this feature with a2billing. This is difficult to extract the working code from a2billing. Also we are developing billing system so this is not a good idea to deploy another billing system in parallel. Any idea or link might help full. On Fri, Sep 24, 2010 at 9:30 PM, Tarek Sawah wrote: A quick answer? A2billing. It has what you call it differential billing.. but they call it progressive billing.. 3 steps .. for 3 different rates .. Go for it.. easy to setup and quick to learn and use. Regards From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 24, 2010 4:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] differential billing From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit Sent: Friday, September 24, 2010 8:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] differential billing Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 Since the CDR records the call duration in seconds, this should be a relative “no-brainer”, assuming you are billing post-call. If you are wanting to generate the charges during the live calls, AMI would be your best option for getting a running duration of the connection. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Abdul Basit | +92 32 1416 4196 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] differential billing
the way i see it can be done.. is using ${CDR(billsec)} into a dial plan or
your AGI, A2billing is a script that runs and waits till the call ends then
exists with status 0. it doesn't listen to AMI (as i expect) it pulls the
variables after the channel is hungup and then does the calculation something
like
ROUND(${CDR(billsec)}/60) to get FULL MINUTES if calculation in full minute
round for example then do the calculation in mysql when inserting the
sessionbill or let the PHP or AGI script do the math for you.
200 calls won't be a problem with Server that has good resources.
Just give it a try and let me know.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
From: basit.e...@gmail.com
Date: Sun, 26 Sep 2010 02:43:05 +0500
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] differential billing
Yes. you are right. I was thinking to avoid reinventing the wheel.
Will write AGIs. Trick is how to charge at 3min 59 sec or 4 min 01 sec
during live call.
We can monitor channel variables over AMI. But this will be a CPU
overhead (say for 100 or 200 calls) if we monitor channel variables on
every second. I want some thing to push channel details on each
transition (or events like IVR level changed, call duration updated to
next minute) rather than i request on AMI. Don't know if this logic is
workable.
Just want a right direction.
--
Regards,
Abdul Basit | +92 32 1416 4196
On Sat, Sep 25, 2010 at 11:37 PM, Tarek Sawah
wrote:
if you are deploying your own system.. then you can build a small
application (AGI) that would do the math for you .. will devide the
call duration into the stages you want .. and does the calculation.. i
think MYSQL already can do that.. but a PHP script will do it faster
and easier.. or like our billing system.. C# application interacting
with Asterisk doing all the math. after all it's all SQL and Asterisk
working. you can do that with a dial plan i believe.. so why not build
an AGI to do it for you?
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +13864929993
From: basit.e...@gmail.com
Date: Sat, 25 Sep 2010 23:27:56 +0500
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] differential billing
Tarek,
I already tested this feature with a2billing.
This is difficult to extract the working code from a2billing.
Also we are developing billing system so this is not a good idea
to deploy another billing system in parallel.
Any idea or link might help full.
On Fri, Sep 24, 2010 at 9:30 PM, Tarek Sawah
wrote:
A quick answer? A2billing.
It has what you call it differential billing.. but they call it
progressive billing.. 3 steps .. for 3 different rates ..
Go for it.. easy to setup and quick to learn and use.
Regards
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Danny Nicholas
Sent: Friday, September 24, 2010 4:19 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] differential billing
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Abdul Basit
Sent: Friday, September 24, 2010 8:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] differential billing
Hi All,
How can we develop a differential charging setup using asterisk like
for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next
30 sec charge @15cent, etc?
Any idea, suggestion.
--
Regards,
Abdul Basit | +92 32 1416 4196
Since the CDR records the call duration in seconds, this should be a
relative “no-brainer”, assuming you are billing post-call. If you are
wanting to generate the charges during the live calls, AMI would be
your best option for getting a running duration of the connection.
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Abdul Basit | +92 32 1416 4196
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Re: [asterisk-users] Fax On Demand - Asterisk 1.4.29
i don't see any mistakes in your question.. but i still don't get it.
what do you need exactly from Fax on demand? sending faxes? receiving faxes?
From: zoelha...@yahoo.co.id
To: asterisk-users@lists.digium.com
Date: Fri, 24 Sep 2010 17:27:57 +0700
Subject: [asterisk-users] Fax On Demand - Asterisk 1.4.29
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Hi All,
Is there anyone who ever implemented successfully Fax On Demand
on Asterisk 1.4.29 ?
I’ve tried to look from Google about this issue and could not
find any satisfying about this.
Thanks in advance for all of you who willing to help
And Sorry if there’s any mistake in my question, cause this is
my first time asking question in this mailing list.
Thanks
Regards,
Zoel Hairi
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Re: [asterisk-users] differential billing
A quick answer? A2billing. It has what you call it differential billing.. but they call it progressive billing.. 3 steps .. for 3 different rates .. Go for it.. easy to setup and quick to learn and use. Regards From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 24, 2010 4:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] differential billing _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit Sent: Friday, September 24, 2010 8:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] differential billing Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 Since the CDR records the call duration in seconds, this should be a relative no-brainer, assuming you are billing post-call. If you are wanting to generate the charges during the live calls, AMI would be your best option for getting a running duration of the connection. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] differential billing
A quick answer? A2billing. It has what you call it differential billing.. but they call it progressive billing.. 3 steps .. for 3 different rates .. Go for it.. easy to setup and quick to learn and use. Regards From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 24, 2010 4:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] differential billing _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit Sent: Friday, September 24, 2010 8:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] differential billing Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 Since the CDR records the call duration in seconds, this should be a relative no-brainer, assuming you are billing post-call. If you are wanting to generate the charges during the live calls, AMI would be your best option for getting a running duration of the connection. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Digium TC400B
Greetings, Because of the heavy load and the high expectations of an asterisk server offered as a solution integrated with our CRM software.. we were looking into other possibilities than software Licenses for G729 and G723 codecs.. to lower the pressure on the processor giving it more space to do more work. We heard of a hardware (PCI CARDS) can be used with Asterisk that does the work. And we stumbled with Digium TC400B. Could be a newbie's question.. but does that serve our needs? As we have not pressured a server before up to 1400 extensions with 600 outbound SIP calls (customer's needs). The server in question is Core I7 16 GB ram and Raid 10 SAS drives. We need to know how many calls with G729 or G723 can this server handle? And as far as we can see this Digium card can be a cheaper solution If calculating the CPU cost plus the licenses for each channel. One more question.. can we add two of those cards to the server? Will it be efficient? Regards Tarek Sawah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realm: security issue
Bilal, If you are using 3G or Wifi with your Nokia Native SIP Client.. try to connect via an internet connection sharing machine.. it seems that your ISP is blocking INBOUND SIP packets. Test and let me know -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, September 23, 2010 11:24 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] realm: security issue No, I do not think that my provider blocked my IP address, because I am able to register for the Asterisk (at that IP address) from an IP Phone, but not from the mobile. It is well known that the mobile use the digest authentication (realm) which is not used in the IP Phone. Any advise? From what you explained it seems to me that your mobile provider has blocked your sip communication altogether. Have you tried changing IP address of your asterisk server? If changing IP works, then probably your provider has blocked you sip communication by IP only. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-23 7:22 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I have my friend that use his mobile (Nimbuz) to connect for the Asterisk and his account was working fine. Suddenly it stop working (not able to register). From my mobile (Nokia) I was able to register using my username and password, so I tried to register using his (my friend) username and password (that was using them from Nimbuz), it did not work. I come back trying to register using my origin username and password (which was working fine just before a while), it did not work. I removed my username and my friend username from the Asterisk and then I created a new username and password (different than all other) and I tried to register from my mobile, also it did not work !!! I start beleive that it is something related to detecting a hacking (maybe Nimbuz does not use a good security), this caused the MAC to be considered as hacked. Please, can someone advise me how to resolve this problem? Where I can find those MACs that need to be removed from block list? What can I do to get out from this problem? Any advise? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Gareth
Usualy the queue has the ability to know if the agent is INUSE and skip
them.. you can simply use ringinuse=no to the queues.conf under the queue
itself or the general section and that's it .. no need for the whole
dialplan.. as you are using SIP members.
Salam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Wednesday, September 15, 2010 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Yes something like this. Note the Execif syntax I have used is for
asterisk 1.6
exten = s,n,Set(AGENTSBUSY=yes)
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1009} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1010} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[$AGENTSBUSY = no]?QUEUE(xxx))
Shariq Khan wrote:
You mean, I need to check the DEVICE_STATUS of both (sip) users before
sending the caller into queue, otherwise skip the caller from going into
Queue by using ExecIf.
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 3:16 PM, Gareth Blades
list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com wrote:
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is busy in
the Queue.
I have two channel member in queue and i have set the peer limit
2 for
these members.
I want to skip those member who are currently on the call
(answered to
calls) and now their status is busy, if Queue see the busy status
caller
will not enter in the Queue and go to the next priority.
[test-queue]
strategy = rrmemory
memberdelay=0
timeoutrestart = no
joinempty = strict
leavewhenempty = yes
timeout = 50
member = SIP/1009
member = SIP/1010
sip.conf
[1009]
username=1009
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
[1010]
username=1010
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
--
Regards,
Shariq Khan
0333-3501125
You could use ${DEVICE_STATE(SIP/1009}. Set a variable to indicate all
extensions are busy and then a couple of ExecIf calls to reset the
variable if either of the extensions state is set to NOT_INUSE. You
then
have a variab you can use to decide where to jump to in the dialplan
depending on whether both phones are busy or not.
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Re: [asterisk-users] SIP softphones answer but do not connect...
can you state your internet connection your agents are on?and one more thing.. how are the members positioned into the Queue? static? Dynamic? single station and call forwarding (find me follow me extension in the queue)? do you get call waiting override with Auto Answer? -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 From: cur...@telecomabmex.com To: asterisk-users@lists.digium.com Date: Mon, 13 Sep 2010 10:44:35 -0500 Subject: Re: [asterisk-users] SIP softphones answer but do not connect... On Mon, 2010-09-13 at 12:49 +1200, Matt Riddell wrote: On 11/09/10 12:44 PM, Carlos Chavez wrote: The past few days I started having a problem with a small call center setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam is configured to auto answer the call. The problem is that the agents claim that they get a call but no audio. From the logs I can see that it is calling the agent phone but after 10 seconds (the queue timeout for pickup) I get the message that nobody answered and the call is sent to the next available agent. This can happen with up to three agents (the third finally answers the call). This has happened at least 20 times in the past two days. At first the supervisor thought that the same call was ringing on three different agents at once but the logs say that the first two do not answer and the third does. What strategy are you using for the Queue? We are using Least Recent at the moment. Why would queue strategy impact this? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A way to check against a list of numbers?
i have this scenario where i have a marketing department calling USA numbers
excessively and sometimes the leads contain duplicate numbers OR duplicate
customers with different numbers on the other hand we have some numbers that
are black listed the destination should be checked and caller should be
informed in both cases.
the following dialplan would first check if the number is blackliste (from
local MYSQL DB) .. challenge it then continue to MSSQL DB where existing
customers info is located and challenge the phone number against existing
customers to see if the call should go through or not.
exten = _1N.,1,MYSQL(Connect connid localhost localSQLuser password
blacklistDB)exten = _1N.,n,MYSQL(Query resultid_1 ${connid} SELECT COUNT(*)
FROM tbl_BlackList WHERE PhNumber=${EXTEN})exten = _1N.,n,MYSQL(Fetch fetchid1
${resultid_1} ifpresent)exten = _1N.,n,MYSQL(Disconnect ${connid})exten =
_1N.,n,GotoIF($[${ifpresent} = 0] ?pok:perror);;; IF THE NUMBER EXISTS TELL THE
CALLER THAT IT'S BLACKLISTEDexten = _1N.,n,MYSQL(Clear ${resultid_1})exten =
_1N.,n,MYSQL(Clear ${fetchid1})exten = _1N.,n(perror),Wait(1)exten =
_1N.,n,PlayBack(privacy-blacklisted)exten = _1N.,n,congestion(1)exten =
_1N.,n,HangUpexten = _1N.,n(pok),GoToIf($[${ODBC_CHKAVAIL(${EXTEN})} =
0]?dial:exerror)exten = _1N.,n(dial),GoTo(dial-usa,${EXTEN},1)exten =
_1N.,n(exerror),PlayBack(already-in-db) ;;; PLAY SOUND FILE THE CUSTOMER
ALREADY IN DATABASEexten = _1N.,n,Hangup
you can use the above example to check the number being dialed against your DB
(what ever DBMS you are using) and route it depending on the result of your SQL
query.hope this helps
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
From: benny+use...@amorsen.dk
To: hose+aster...@bluemaggottowel.com
Date: Mon, 13 Sep 2010 20:18:08 +0200
CC: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] A way to check against a list of numbers?
Hose hose+aster...@bluemaggottowel.com writes:
The most straightforward way would be to just define explicit patterns.
Obviously that works, but doesn't seem scalable in terms of maintenance.
I don't think that maintaining the list in the dial plan is all that
bad, actually. Dump it in its own context and file...
If that isn't convenient enough I'd go for the Asterisk database next.
Also on the option list is private e164/enum or an SQL database.
/Benny
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Re: [asterisk-users] SIP Delay with remote stations?
this can be cause if you are using an ADSL link with your remote phones .. or maybe some 3G networks can cause that delay in the first response as the ACK message will be late to arrive and if the delay was too high .. the call will drop.one more thing if your remote phones are (Queue Members) this can be caused by a configuration of the queue itself something related to memberdelay directive. try setting it to 0 or something similar.Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993 From: william.stillwell-li...@ablebody.net To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 10:06:55 -0400 Subject: [asterisk-users] SIP Delay with remote stations? I have several remote phones that experience a slight “call” delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. _ The New Busy think 9 to 5 is a cute idea. Combine multiple calendars with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multicalendarocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_5-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hot to configure trunk in asterisk with a2billing.
Lets say you did everything as it was mentioned in the tutorial .. then go into Asterisk console and issue the command:sip show peer A2BILLINGCREATEDUSER if you can't find it.. then simply include additional_a2billing_sip.conf in your sip.conf file.Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Tue, 29 Jun 2010 13:41:22 +0530 From: alagudr...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hot to configure trunk in asterisk with a2billing. Hi All, I am newbie in this asterisk and a2billing technology . i had successfully installed asterisk in my server fedora -8 [server behind NAT/STUN] i after installation i can able to create users and tested the call features with X-Lite . the was working fine . after i installed the A2Billing in my same server with follow the steps from a2billing installation guide. but u cant access the users from a2billing in asterisk . if i am trying to access the username which is created in a2billing it displayed request timeout somewhere i missed the configuration, please help me to resolve this error . Thanks, Gokul. _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restricting sip users to a certain useragent
well there are two restrictions.. the IP address of the station they are using it .. and the UserAgent..one thing my agents hardly understand Computers .. and their computer skills are limited to Microsoft Office products and telemarketing. i'm not afraid of hackers or cracker .. security is not guaranteed .. but i need to restrict the agents to their seats and my CRM software -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 08:45:01 +0100 Subject: Re: [asterisk-users] restricting sip users to a certain useragent On Tuesday 29 Jun 2010, Tarek Sawah wrote: . is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will prevent the agents we are restricting them to only be able to dial through the software which is already restricted to their seats in the call center.. but someone might sniff around .. and get the sip username and password assigned to him and use it through Zoiper or any other softphone to make calls . If someone is *that* determined, what will stop them from modifying the user-agent string in some Open Source softphone? -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] restricting sip users to a certain useragent
Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM Customer Relations Management with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will prevent the agents we are restricting them to only be able to dial through the software which is already restricted to their seats in the call center.. but someone might sniff around .. and get the sip username and password assigned to him and use it through Zoiper or any other softphone to make calls ..our agents are allowed international calls .. so we want to restrict them to only use our dialer.Is that possible?Asterisk version 1.4.33regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big time system
a Rack of load balanced Asterisk Servers with some customized billing system with a respectable centralized database like MsSQL or Oracle ..External E1 or T1 Gateways instead of TDM cards.. with load balancing?? as the whole operation is COPPER WEIRES .. can't that setup work for them?I'm asking as i'm looking for a similar setup just trying to set it up virtually before we go live.Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Fri, 25 Jun 2010 11:49:12 -0400 From: j...@ngn-networks.com To: asterisk-users@lists.digium.com; ca...@usawide.net Subject: Re: [asterisk-users] Big time system Cary- Asterisk may carry you a way down this road, but in the end, it's not, and was never designed to be a class 5 telecom switch. There are people working on a carrier grade implementation that may or may not be fully class 5, but I don't know what the status is on that. I haven't gotten an answer from Digium on that lately. What you're looking for are local gateways that backhaul to a central switch site with equipment that can support traffic from multiple rate centers in multiple LATAs. This gets complicated quickly, especially if your rate centers are spread across multiple states. You'll want some type of Multiservice Access Platform (MSAP). Zhone makes the MALC and their newer MXK box. Adtran has the TA-5000 shelf. Neither are what you'd call cheap. Both will provide T1 access, DSL, SDSL, VDSL, bonded, and even ethernet access to the customer over a variety of transport options, including copper pairs. The Zhone box already has SIP backhaul for voice traffic, and the Adtran shelf should have it soon. Today the Adtran box has GR303 backhaul for voice. All that said, what you're proposing indicates to me that you're likely to need to establish CLEC certification in whatever states you'll be operating. That in itself is not a short process. It can take anywhere from 90 days to a year depending on the state, and expect to spend from $10K up on legal costs per state alone. Insurance, financial health, and other requirements vary by state as well. The ILECs generally won't even talk to you about establishing colo and gaining access to the copper loops until you get the CLEC certificate. Generally the process starts by getting the certificate, then negotiating an ICA, then trunking services, then colo. Different carriers will be easier to work with than others, but they are all a pain. ATT requires you to have a $10M general liability policy in place before you can even submit a request for a space availability report. All this is not to say it can't be done, but to point out that it's a very difficult process to negotiate, even when you have done it several times. Without experience it can be close to impossible. I'd suggest getting a good telecom/clec consultant and a good telecom lawyer (I know a few) involved early in the process, or you'll end up spending ALOT of money. Hit me off-list and I can give you more info. Joe On 6/24/2010 11:24 PM, Cary Fitch wrote: We are an asterisk user... small time system 50-100 users or so. But, we have an opportunity to get into a big time telecom activity. It would have 2000 to 30,000 user lines per city, and we would like to have those brought back to a central location for control and because transport can be more economical than remote site rentals, maintenance and personnel. We could take the local lines into concentrators (TNTs or equivalent) and bring back IP to a central site, or put servers at the remote cities. Our object is to serve as a central office switch for subscribers on standard telco service loops. This isn't a How many lines can I handle using a Belchfire 2600 processor? type question but a request for pointers to big time systems. There would be no IP path to the end user, just copper. Thank you Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] Need USA DIDs
didforsale.com is one of the best and reliable DID providers in the USA -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Wed, 23 Jun 2010 16:50:48 +0530 From: rscl.mum...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Need USA DIDs Hi, Looking for some reliable and quality providers of USA DIDs. Any pointers ? Thx Sans _ The New Busy think 9 to 5 is a cute idea. Combine multiple calendars with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multicalendarocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_5-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with sip 401 unauthorized
i faced a similar situation with my ISP .. they block INBOUND UDP port 5060 which means if i try to register.. the server would receive my registration message.. but when it sends the acknowledgement .. the ISP Firewall rejects the message so the server responds with Unauthorized.. i simply changed the port on the server to 5070 and set my dialer to listen to port 5070 as well (for inbound messages) and this solved my issue.that was my situation.. so your problem is in the firewall settings.. just try to look at it and see what is missing.. and by the way when you send all of your IP sections XXX no one will assist you as no one will know who is talking to whom.. just like if you go to a doctor with a prostate problem.. you can't tell him that you won't remove your clothes off ;)regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993 Date: Wed, 23 Jun 2010 08:44:21 -0400 From: ge...@pagestation.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] help with sip 401 unauthorized I am getting a SIP 401 unauthorized message. My public IP or PIP is being pre-routed with iptables to goto an internal IP or IIP All the polycom phones in the office point to the IIP. they work fine. I have 2 external phones that are registering to the PIP. I see the register attempt as I am getting the 401 unauthorized message. For the 2 external phones both have nat=1 enabled. remote phone (192.X.X.X) GW internet PIP (prerouted) (74.X.X.X) internal server (192.X.X.X) This used to work before I moved the server inside the firewall. What special setting do I need to enable to get this working. Thanks, Jerry --- Transmitting (NAT) to X.X.X.X:1024 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK6ea01bc7;received=X.X.X.X From: sip:x...@x.x.x.x.;user=phone To: sip:x...@x.x.x.x;user=phone;tag=as21ab1732 Call-ID: 000ff78d-ebb20007-22675f66-5da7e...@x.x.x.x CSeq: 1196 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1c6a6002 Content-Length: 0 [XXX] type=friend username=XXX secret= dtmfmode=RFC2833 host=dynamic context=external rtptimeout=60 qualify=no canreinvite=yes nat=yes disallow=all allow=ulaw allow=alaw allow=gsm -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one for your filters
you can start by simply telling us what is the purpose of your server.. and does it have long distance of overseas?? do you use Numeric usernames? simple passwords? passwords the same as your username? this way you can offer more info so we can help you.a quick answer will be.. opening a few and blocking ALL is easier.. as you can have upto 400 prefix to block .. unless you call world wide.. then you will have to block the countries you don't call .. another option.. make your usernames more complex.. letters and numbers.. an additional option is to use fail2ban with Asterisk support.. it will block the IP after the number of attempts you set in the configs. a client of mine wanted simple usernames and passwords to be setup using the keypad on the ipphones.. two months ago they had the same problem you faced.. 400$ to Zimbabway .. and later on 1200$ to Zimbabway.. their provider have a limit of 30 minutes per call .. so the caller had to redial.. unless it's automated.still you can provide us with more info.Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993 Date: Wed, 23 Jun 2010 16:08:51 + From: j...@sunfone.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] one for your filters Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place four thousand calls to what appears to be a toll number in Zimbabwe last night. Filter 82.150.165.5. A more overriding problem for me is how do we know what *destinations* to filter so this idea of war dialing a toll number is something we can cutoff before it gets to our upstream provider? Is there some collected list of toll prefixes that I can filter on? Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need USA DIDs
i consuleted didforsale.com regarding the wholesale thing and their response was that you should buy a bulk of numbers and make your own api.. one more thing.. if you are in the USA ..be sure to start your FCC registration (if you don't have it yet) because it can be a disaster for US companies providing DID numbers to US citizens without FCC license. -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Wed, 23 Jun 2010 23:43:14 +0530 From: rscl.mum...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Need USA DIDs On Wed, Jun 23, 2010 at 9:50 PM, Hall, Rick r...@readywire.com wrote: Agreed! Didforsale.com is THE way to go. -- Rick Hall Senior Vice President ReadyWire Multimedia Solutions Anyone having experience with didww.com ? Sorry, I forgot to mention I am looking for wholesale DID -- reseller option with API to that my customers can select country - city -- DID from my website. Thx _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk distribution for a Call Center
i have been struggling with call center Customers for a couple of years now.. i have a call center with 40 agents using elastix.. and quality is related to the source of calls inbound or outbound... the problem with call centers they need Visual .. like Flash Operator panel and CDRs.. if you can go with simply raw asterisk .. without any additions.. will be the best for you .. write your own dial plans.Flash operator Panel is not a flawless work.. and adds more burden on the resources.. esp when it's open by 7-8 persons at once.. regarding the ACD ..it's all about PHP and Database .. you can build your own reports and so. or you can use a2billing to do the billing and ACD.. Elastix has a good billing (without a2billing) .. but i prefer a clean installation of asterisk and work around with database and PHP much better.. Good Luck! -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 (386) 492-9993 Date: Tue, 22 Jun 2010 15:21:18 -0300 From: aco1...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk distribution for a Call Center Dear all, I need to build a PBX based on Asterisk for a call center. I have worked with raw Asterisk but it's hard to work for big implementations think. Also I have worked with Trixbox CE for a small bussines and it was prette good, but I have not have many features like ACD. I know there is another version called Trixbox PRO -specially Call Center edition- that's not free but has got more features like ACD and billing. I've heart about AsteriskNow and I know it's free. What distribution/version do you recommend to me in order to implement a call center and taking into account I'm not an expert in programming from Asterisk CLI ??? Thanks a lot Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
what do you mean unblock the calls exactly? -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Fri, 18 Jun 2010 11:12:55 +0100 From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk issue Hello, I have a problem in Asterisk 1.4 each day I need to restart asterisk service asterisk restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to stop intruder from registering sip?
along with all the previous suggestions.. i found out that fail2ban is a good safe tool to be used along with hard passwords and not using numeric usernames.. for me using A2Billing along with Asterisk was a pain because it needs to create usernames numeric.. so i had to create strong SIP users and passwords then assign a2billing accounts to them to make it safer.. plus the fail2ban .. give it a try. -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Sun, 13 Jun 2010 22:28:38 -0700 To: asterisk-users@lists.digium.com From: i...@extrasensory.com Subject: Re: [asterisk-users] How to stop intruder from registering sip? At 01:06 PM 6/13/2010, you wrote: We use a combo of aastra 9133i and 57i's. Don't the user id and the extension HAVE to be the same? I had thought the aastra's used the extension as the SIP id to register. So in your extensions.conf you need lines like: exten = 123,1,dial(SIP/123_thisisAfunnyextension) Well, that should give you the idea. Don't know if it's the best way, but it's worked for me. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call queues - issues, can't make it work.
when you add an agent to a queue the agent should log in try adding
member=SIP/301member=SIP/302instead of agent directives.this will ring both
phones.. from your output it doesn't seem to be ringing the agents at all.
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
From: ak...@abacus-it.no
To: asterisk-users@lists.digium.com
Date: Mon, 14 Jun 2010 13:41:20 +0200
Subject: [asterisk-users] Call queues - issues, can't make it work.
Hello there
I have been struggling with queues, because
i think this is the right module for our business.
My main goal, is when we receive external
calls, the receptionist should be able to transfer the call to us
Technicians, and I am trying to add 2
extensions to a queue name [teknisk]
Extension 301 and 302.
I have a test setup now which I thought
should look like this:
When a external call come to my external
number (67209611) this will ring for 5 seconds, and then transferred to queue
“teknisk”
And I thought that internal
phonex/extensions 301 and 302 would ring.
But, when I ring the external number, it
just rings…and rings…until it hang-ups.
CLI output shows that the commands are
running, but maybe the wrong way, are the queue command routed to my sip
provider?
Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in
technician department, which I am trying to route the queue to with the ringall
argument.
This happens:
Reloading MGCP
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing
[4767209...@internal:1]
NoOp(SIP/odin.service.ipallover.net-00d1, ) in new
stack
-- Executing
[4767209...@internal:2]
Verbose(SIP/odin.service.ipallover.net-00d1, Callerid num
90015103) in new stack
Callerid num 90015103
-- Executing
[4767209...@internal:3]
Dial(SIP/odin.service.ipallover.net-00d1,
SIP/301,5) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is
ringing
-- Nobody picked up in
5000 ms
-- Executing
[4767209...@internal:4]
Queue(SIP/odin.service.ipallover.net-00d1, teknisk)
in new stack
-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1
--
SIP/odin.service.ipallover.net-00d1 Playing 'queue-youarenext.gsm'
(language 'en')
-- Told
SIP/odin.service.ipallover.net-00d1 in teknisk their queue position (which
was 1)
--
SIP/odin.service.ipallover.net-00d1 Playing 'queue-thankyou.gsm'
(language 'en')
-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1
== Spawn extension (internal,
4767209611, 4) exited non-zero on 'SIP/odin.service.ipallover.net-00d1'
asterisk*CLI
---
Agents.conf is default and i have two
extensions/agents
agent = 301,301
agent = 302,302
--
[r...@asterisk asterisk]# more queues.conf
[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes
member = Agent/301
member = Agent/302
-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped
;exten 301
exten = 4767209611,1,NoOp();
exten = 4767209611,n,Verbose(Callerid
num ${CALLERID(num)});
exten = 4767209611,n,Dial(SIP/301,5);
exten = 4767209600,n,Queue(teknisk);
exten =
4767209611,n,Voicemail(301);
;Added 06.Mai.10-Aksel
Could someone please help me in the right
direction here?
Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner
Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no
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Re: [asterisk-users] calling peer from server
does that phon has a static IP? does it register with the server? posting your SIP.con and extensions.conf related to this issue could help us to understand what you are doing. -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 From: niksingha...@gmail.com Date: Mon, 14 Jun 2010 17:49:37 +0530 To: asterisk-users@lists.digium.com Subject: [asterisk-users] calling peer from server Hi everybody, This is the console output of the asterisk server. debian-te410*CLI sip set debug peer 2002 SIP Debugging Enabled for IP: 172.26.48.113:5061 I have a sofphone with user 2002 registered on the server on the ip 113. I am trying to place a call to the sofphone on this ip. I have written a simple php script which utilises the exec_dial function inbuilt in phpagi.php file. I have tried diff ways but can't seem to get it work. Can please some one suggest me anything in this regard. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue ringall problem.
a portion of your quues.conf and you sip.conf pasted can be helpful? try using autofull=yes in your queues.conf and see if it works -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Mon, 31 May 2010 11:33:09 +0200 From: mass...@archivio.it To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue ringall problem. This is the problem: Call coming into a queue in ringall strategy, if a member (SIP) of the queue is busy when entering the queue, and this member comes free after a little time, the member never rings.. How to solve this? I changed all parameters of the queue with no results... Wath i need: If one member of the queue is busy when a new call come in to the queue, this member can hangup and try to answer the new call Thnks. _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue ringall problem.
it's autofill=yes i'm sorry for the typo -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Mon, 31 May 2010 11:33:09 +0200 From: mass...@archivio.it To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue ringall problem. This is the problem: Call coming into a queue in ringall strategy, if a member (SIP) of the queue is busy when entering the queue, and this member comes free after a little time, the member never rings.. How to solve this? I changed all parameters of the queue with no results... Wath i need: If one member of the queue is busy when a new call come in to the queue, this member can hangup and try to answer the new call Thnks. _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing DID and Queues
the simple way i can see it is the following;let's say you have did starts with 1708 [from-did]exten = _1708XXX,1,Answerexten = _1708XXX,n,Queue(SALES,,)exten = h,1,Hangup -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Tue, 18 May 2010 20:47:12 -0700 From: toqee...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] a2billing DID and Queues Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one extensions does not receive the call so it should be forwarded to the next extensions. So please help, Any help will highly appreciated. Thanks -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916 _ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and RFC 3261
Greetings List,Trying to interconnect with a new provider.. the require a compliance with RFC 3261 so knowing less than needed about RFC documentations.. i would like to know if Asterisk is actually in compliance with RFC 3261 or not.. Can any one help with this? Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
i'm having the same problem with one of my call centers located in Egypt.. although the ip-phones are located on a Dedicated Leased Line yet calls drop out of the blue.almost an identical setup as yours..provider located in France (data center) my server located in Sweden (data center) both on public network no NAT.. and the remote office is behind NAT.somehow i suspect Internet problems with your case.. as RTP packets should not stop arriving unless internet connection is timing out. i suppose your calls that are dropping are INBOUND coming from your provider and directed to your remote location.. and you don't have any problems with OUTBOUND calls from your remote location to your server ( I have setup a loop test that goes between 5 locations originating from my remote location and returns to the remote location through 5 hops including IPKALL servers and call goes well with no problem). and let me take a wild guess.. your provider is offering a premium number services.my advise check your internet connection on the remote location and keep a ping from that network to your server running all the time to check for time outs. -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 From: d...@keshercommunications.com To: asterisk-users@lists.digium.com Date: Thu, 29 Apr 2010 16:33:06 -0400 Subject: [asterisk-users] Calls Dropping Hi, I’m having a major problem with random calls dropping. After spending weeks trying to figure it out, i’ve finally spotted the issue but don’t know how to resolve it. I run a sip server that’s hosted in a data centre. It has a public IP address with no nat involved. My provider also has a public ip with no nat involved. The sip phones are in a remote office behind a nat router. Every so often, all the rtp data coming from the remote location stops arriving at my sip server. So after about 30 seconds, the call gets terminated by my provider because i’m not sending any rtp packets to them. Any ideas why the rtp data should stop coming in, and how can I resolve it? Asterisk v1.4.30 6 x Linksys SPA921 Router at remote site is a Thomson TG585v7 Any assistance will be greatly appreciated. Many thanks Dan _ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_2-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Invite issue
Before posting let me mention that this doesn't happen with ALL destination on this provider.. some destination doesn't face this problem .. but this is a sample call [K -- Executing [0020100324...@a2billing:1] [1;36;40mDeadAGI[0;37;40m([1;35;40mSIP/58169-ac47fda0[0;37;40m, [1;35;40ma2billing.php|1[0;37;40m) in new stack [K -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- AGI Script Executing Application: (Dial) Options: (SIP/PROVIDER1/20100324519|60|HL(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 start_sound = (null) warning_sound = timeleft end_sound = (null)Audio is at 100.X.Y.Z port 13984Adding codec 0x100 (g729) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 195.X.Y.Z:5060:INVITE sip:20100324...@195.x.y.z SIP/2.0 Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z Contact: sip:58...@100.x.y.z Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Apr 2010 18:52:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 267 v=0 o=root 12516 12516 IN IP4 100.X.Y.Z s=session c=IN IP4 100.X.Y.Z t=0 0 m=audio 13984 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called PROVIDER1/20100324519 [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Content-Length: 0 - [K --- (7 headers 0 lines) --- [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: sip:20100324...@195.x.y.z:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 260 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z s=SIP Media Capabilities c=IN IP4 195.219.240.5 t=0 0 m=audio 15846 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=maxptime:20 - [K --- (11 headers 12 lines) --- [K Found RTP audio format 18 [K Found RTP audio format 101 [K Peer audio RTP is at port 195.219.240.5:15846 [K Found audio description format G729 for ID 18 [K Found audio description format telephone-event for ID 101 [K Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [K Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [K Peer audio RTP is at port 195.219.240.5:15846 [K -- SIP/PROVIDER1-1fd586a0 is ringing [K -- Call on SIP/PROVIDER1-1fd586a0 placed on hold [K -- Started music on hold, class 'default', on SIP/58169-ac47fda0 [K -- SIP/PROVIDER1-1fd586a0 is making progress passing it to SIP/58169-ac47fda0 [K sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 195.X.Y.Z 2010032451 7f169cce700 00102/0 0x100 (g729) Yes Init: INVITE 78.184.197.119 58169 AC8455D8edd 00101/160518 0x4 (ulaw) No Rx: INVITE 2 active SIP channels [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: sip:20100324...@195.x.y.z:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 0 - [K --- (9 headers 0 lines) --- [K -- SIP/PROVIDER1-1fd586a0 is ringing -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Thu, 29 Apr 2010 16:52:24 +0100 From: list-aster...@skycomuk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange Invite issue Can you post a sip debug Tarek Sawah wrote: Greetings List. I'm facing a strange issue with one of my
Re: [asterisk-users] Strange Invite issue
then why is it happening on a few destinations on that particular provider? Date: Fri, 30 Apr 2010 13:09:05 -0700 From: david.wh...@watchguard.com To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange Invite issue in the SIP/2.0 180 Ringing, the SDP shows: a=sendonly this is hold by rfc 3264. then when the other end picks up, a new SDP is probably sent with a=sendrecv I believe your server is acting correctly. -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Tarek Sawah Sent: Fri 4/30/2010 12:11 PM To: Asterisk Users Subject: Re: [asterisk-users] Strange Invite issue Before posting let me mention that this doesn't happen with ALL destination on this provider.. some destination doesn't face this problem .. but this is a sample call [K -- Executing [0020100324...@a2billing:1] [1;36;40mDeadAGI[0;37;40m([1;35;40mSIP/58169-ac47fda0[0;37;40m, [1;35;40ma2billing.php|1[0;37;40m) in new stack [K -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- AGI Script Executing Application: (Dial) Options: (SIP/PROVIDER1/20100324519|60|HL(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 start_sound = (null) warning_sound = timeleft end_sound = (null)Audio is at 100.X.Y.Z port 13984Adding codec 0x100 (g729) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 195.X.Y.Z:5060:INVITE sip:20100324...@195.x.y.z SIP/2.0 Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport From: 58169 ;tag=as00522e07 To: Contact: Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Apr 2010 18:52:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 267 v=0 o=root 12516 12516 IN IP4 100.X.Y.Z s=session c=IN IP4 100.X.Y.Z t=0 0 m=audio 13984 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called PROVIDER1/20100324519 [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 ;tag=as00522e07 To: ;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Content-Length: 0 - [K --- (7 headers 0 lines) --- [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 ;tag=as00522e07 To: ;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 260 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z s=SIP Media Capabilities c=IN IP4 195.219.240.5 t=0 0 m=audio 15846 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=maxptime:20 - [K --- (11 headers 12 lines) --- [K Found RTP audio format 18 [K Found RTP audio format 101 [K Peer audio RTP is at port 195.219.240.5:15846 [K Found audio description format G729 for ID 18 [K Found audio description format telephone-event for ID 101 [K Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [K Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [K Peer audio RTP is at port 195.219.240.5:15846 [K -- SIP/PROVIDER1-1fd586a0 is ringing [K -- Call on SIP/PROVIDER1-1fd586a0 placed on hold [K -- Started music on hold, class 'default', on SIP/58169-ac47fda0 [K -- SIP/PROVIDER1-1fd586a0 is making progress passing it to SIP/58169-ac47fda0 [K sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 195.X.Y.Z 2010032451 7f169cce700 00102/0 0x100 (g729) Yes Init: INVITE 78.184.197.119 58169 AC8455D8edd 00101/160518 0x4 (ulaw) No Rx: INVITE 2 active SIP channels [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 ;tag=as00522e07 To: ;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length
[asterisk-users] Strange Invite issue
Greetings List. I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered.. this is happening only with this provide although i have 3 other providers i route calls through.. can anyone explain what is going on? -- Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308 _ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound route question
Simply place the SIP Extension of the GSM gateway in another context context=from-gsm and in your extensions.conf use something like this [from-gsm] exten= = _X.,1,Goto(whatever IVR you want) Date: Mon, 26 Apr 2010 17:23:40 -0300 From: aco1...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Inbound route question Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and 1002) and a GSM Gateway with SIP extension . Two cell phones call to the GSM Gateway number and after that they get a ring tone to dial to the SIP extensions. Is it possible to consider the GSM Gateway SIP extension as an incoming call to the Asterisk PBX and so create an inbound route that point: GSM Gateway DID: - IVR in order to point all incoming cell phone calls to my existing IVR ??? Thanks a lot. Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending RTP media to a different server than SIP Signaling
Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
you got the name EXACTLY! i already am doing what you suggest but facing problems with some destinations and they claim that the problem is with my Asterisk server not their routes! -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sat, 10 Apr 2010 15:50:52 -0400 From: bruceb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling Just a week ago, I have been in the same situation. Provider was changing from Cisco gateways to I think Nextone and hence provided me many IPs. I found out that the media IPs don't matter and just played around with my NAT settings and all calls can go through just fine by using simply: host=111.111.111.111 and the 111.111.111.111 is just their SIP signaling IP. Their gateway will then transfer asterisk to proper gateways for media. Just give it a try; it should work. But my efforts on finding anything regarding this failed on Google as well. P.S. the voip provider name starts with a T and end with A. Regards, Bruce On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah wrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
we started with them two days ago .. and we are facing plenty of False Answer cases on several destinations although ppl said they have a policy against FAS.. anyway i don't know i will be looking into another method to send the RTP to another server, thanks for the info Date: Sat, 10 Apr 2010 18:06:22 -0400 From: bruceb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling Oh, I see. I haven't done a lot of testing on this new IP since the change of gateways happened but I did Canada calls and they go fine. However, this exact provider lies down to their teeth when it comes to problems of call quality and calls not routing. They never accept faults. They even have problems sending calls to Canada and USA. They failed to pass calls to India as well over times. I had a funny issue where they were blocking one specific area code in USA without even telling us. It was just a regular area code. They told me it was blocked but I know it was a lie because they wanted to cover their a$$ as the route was down and it wasn't blocked. I doubt the problem is with sending calls to different media gateway as I think SIP signals take care of that. Just like canreinvite feature. But I reserve the right to be wrong. -Bruce On Sat, Apr 10, 2010 at 4:45 PM, Tarek Sawah wrote: you got the name EXACTLY! i already am doing what you suggest but facing problems with some destinations and they claim that the problem is with my Asterisk server not their routes! -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sat, 10 Apr 2010 15:50:52 -0400 From: bruceb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling Just a week ago, I have been in the same situation. Provider was changing from Cisco gateways to I think Nextone and hence provided me many IPs. I found out that the media IPs don't matter and just played around with my NAT settings and all calls can go through just fine by using simply: host=111.111.111.111 and the 111.111.111.111 is just their SIP signaling IP. Their gateway will then transfer asterisk to proper gateways for media. Just give it a try; it should work. But my efforts on finding anything regarding this failed on Google as well. P.S. the voip provider name starts with a T and end with A. Regards, Bruce On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah wrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list
[asterisk-users] Asterisk for productive Calling Card System
Dear List, i have been thinking of building a calling cards solution based on Asterisk and a2billing.. i have a few questions regarding this solution and was hoping you may have the answers and could be generous enough to offer them. the servers i'm thinking of are with the following Specs: Processor: Intel X3210 Ram: 8Gb HDD: 2x500 GB Sata Internet Link: 100mbps Dedicated was thinking of using one for Database and the other for SIP trunking and calling card purposes. my questions are: 1- from your experience .. would a server with the previous specs handle a pressure of 200 or more outbound calls and 200 inbound (from access numbers)? what are the approximate concurrent call count supported by this hardware? 2- do you suggest using 64bit Centos or other OS? 3- for such usage what codecs do you prefere? 711u? 723? gsm? knowing that we are after good quality calls. my experiences are with small call centers up to 40 seats .. Thank you for your help and support. -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469226/direct/01/ -- _ -- 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] Getting the phone number an SIP extention is dialing
This is the first time i face this issue.. i have an extension 100 .. calling 0018001234567 is there a way in Asterisk to get info that 100 is calling that number? sorry for the lame question but i never had to know such info on my system. -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Hotmail: Powerful Free email with security by Microsoft. http://clk.atdmt.com/GBL/go/171222986/direct/01/___ -- 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] Queues without agent login
Simply use member=SIP/Tarek member=IAX2/JONAS member=LOCAL/whatever simple and good.. with member=SIP/extension i'm facing a CALL WAITING issue.. the agent hears a callwaiting signal whenever the queue tries to call .. so i woul dsuggest using call-limit and busy limite with all your Agents -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: jonas.kell...@telenet.be To: asterisk-users@lists.digium.com Date: Wed, 18 Nov 2009 16:21:12 +0100 Subject: [asterisk-users] Queues without agent login Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to call an extension to log in (and having to give some PIN). I just want a queue (with MoH) to collect multiple incoming calls and then one at a time transfer them to an available SIP-phone. Is this possible ? Thanks you. Jonas. _ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009___ -- 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] SendText
i have my own SMS provider as we sell SMS .. so i have setup my call center with SMS sending for several services and alerts like a Missed Call when i'm not registered it will send me an sms to alert me. it's pretty the same as Matt discribed.. you call an AGI which may use cURL to hit the HTTP API -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Mon, 9 Nov 2009 22:19:08 -0500 From: thomas.per...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SendText Will text messages work to non-SIP enpoints using your logic/code? thank you On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.com wrote: On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 _ Windows 7: Unclutter your desktop. http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen:112009___ -- 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] Termination Question
for the sake of bandwidth you are supposed to connect each two servers together.. otherwise calls between B C will have to go through A . -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: i...@saudihome.com To: asterisk-users@lists.digium.com Date: Thu, 12 Nov 2009 16:13:10 +0300 Subject: [asterisk-users] Termination Question Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. _ Windows 7: Unclutter your desktop. http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen:112009___ -- 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] SIP interconnection problem
you need to post you SIP.conf and your Extensions.conf so someone can have a look at them and see if there is anything missing what are the contexts you are using with your peers? what is the dial plan triggered when calling your destination number? -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sun, 25 Oct 2009 15:19:28 +0100 From: robert.bie...@xponaut.se To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP interconnection problem Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a Failed to authenticate on INVITE on the * to which the Zoiper is registered: -- Accepting AUTHENTICATED call from 192.168.10.113: Zoiper IP requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine -- Executing [010...@users:1] Dial(IAX2/2200-12940, SIP/010...@192.168.10.11) in new stack == Using SIP RTP CoS mark 5 -- Called 010...@192.168.10.11 Other * [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '2200 sip:2...@192.168.10.77;tag=as3e4fedb8' 192.168.10.77 == * for Zoiper -- SIP/192.168.10.11-0a1716f8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION' -- Hungup 'IAX2/2200-12940' Why does * try to authenticate on sip:2...@192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on the IAX phone (not sure this has any meaning in IAX at all) Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ? TIA /R ___ -- 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 _ Windows 7: I wanted more reliable, now it's more reliable. Wow! http://microsoft.com/windows/windows-7/default-ga.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:102009___ -- 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] context does not work
i faced the same problem with callcentric.. when i register i had to add the extension .. like this egister = 1777MYCCID:SUPERSECRET@callcentric.com/1777MYCCID which caused my context to go to the default context and never use the one i already setup.. so removing the extension in the registration string will solve the issue for me.. and i think it will do the same for you. regards -- AHD Tarek Sawah Date: Mon, 10 Aug 2009 12:55:41 +0200 From: patr...@erdbeere.net To: asterisk-users@lists.digium.com Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
Have you tried installing fring? i still like that app .. supports GREAT quality voice over EDGE and GPRS .. plus WIFI and 3G if available.. i tried it with Skype and it's great.. Asterisk and its great Callcentric VoIP provider and it was great.. one thing though i noticed that at some times you will have to dial again for the call to get setup. regards -- AHD Tarek Sawah Date: Thu, 6 Aug 2009 22:59:40 -0700 From: spamsucks2...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform? On Thu, Aug 6, 2009 at 10:15 PM, Alex Balashov wrote: Which generation of the handset are you using? They differ in their processing power and that may account for at least some of it. Alex, this is just an iPod Touch, not even a handset. It doesn't have a mic at all, I had to add one. But using fairly standard debug logic, The mic isn't noisy because it records beautifully. The SIP services all exhibit the same problem Skype works well! So I inculpate the two SIP clients or their configuration. iSip and WeePhone. Although Skype works, it doesn't satisfy the obvious requirement of connecting to my services via SIP. That would allow me to get calls within wifi range on a SIP pbx of my choice. Although I could make calls as well, that is better done with a real phone ;) r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get back to school stuff for them and cashback for you. http://www.bing.com/cashback?form=MSHYCBpubl=WLHMTAGcrea=TEXT_MSHYCB_BackToSchool_Cashback_BTSCashback_1x1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
been testing with Sun VirtualBox and i managed more than 30 extensions on a 2GHz Dual core machine with 1 GB ram for the VBOX.. just not running recodring or encoding .. things went well -- AHD Tarek Sawah Date: Fri, 7 Aug 2009 08:47:03 -0700 From: jlama...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit? Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a feasible extension limit? 20? 50? 100? Any information would be great. Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls Disconnecting out of the blue .. [Renamed]
Greetings again List. I'm facing a strange case with one of the productive Asterisk servers.. i have 3 providers sending traffic to the call center where agents pickup the calls. calls come into the server Queue Agents Last October .. an undersea cable got disconnected placing Egypt and the countries in the region offline.. when internet came back .. the call center located in Egypt had no SIP protocol working.. and we shifted to IAX.. 26 days later SIP started to work again .. but since then calls started to disconnect out of the blue.. we get calls that may last for 45 minutes.. and end normaly .. and we get calls that ring and disconnect the moment the agent picks up been facing a problem with my client as they use the Flash Operator Panel to monitor the call flow through the server and the regualr setup Queue Local users won't work for them as the Flash operator flash offline static agents as online so the client won't know who is on and who is off.. and it's impossible to teach the agents to Login and Logoff the Queue.. so the only solution is the following.. Caller Queue FindMeFollowMe Extension Local SIP extensions this way .. my client is able to monitor the calls and things won't get complicated.. (this is the setup we have been using for 6 months before the problem with the internet occures) since the internet problem and calls are getting disconnected .. out of the blue.. nothing has changed.. and to make sure things are going well .. we moved the server to a Hosting company in California with 10 mb/s connection speed.. (Same Setup that was working well) and still calls get disconnected.. after a lot of problems with the client .. i asked them to change the ISP (my prime suspect was the internet) and finaly they managed to change the ISP .. but the problem is still there.. my server informations are the following Asterisk 1.4.22-3 Uname -a: Linux 2.6.18-92.1.18.el5 sip.conf ;;Agent Sample from Sip.conf [3000] type=friend secret=3000 qualify=yes port=5060 disallow=all allow=g729 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/3000 context=from-internal canreinvite=no call-limit=1 busy-limit=1 ;;Provider's Sample from Sip.conf [50011] type=peer qualify=yes port=5060 pickupgroup= nat=no host=XXX.YYY.ZZZ.NNN disallow=all allaw=alaw allaw=ulaw allow=g729 dial=SIP/50011 context=from-internal canreinvite=no deny=0.0.0.0/0.0.0.0 permit=XXX.YYY.ZZZ.NNN/255.255.255.255 # extensions.conf ;;the provider sends calls to Virtual DIDs (Extensions) in my system which is 8000 exten = 8000,1,GotoIfTime(07:00-16:00|sat-fri|1-31|jan-dec?ext-queues,*8000,1) exten = 8000,n,Answer exten = 8000,n,Queue(8000,t,,,10) exten = 8000,n,Dial(IAX2/6005:6...@backupserver/11) ;; sends the call to a backup server. exten = *8000,1,Answer exten = *8000,n,Dial(IAX2/6005:6...@backupserver/11) the Providers strictly send calls with codec G.729 my agents get best voice quality with G.711u I need your advice .. am i missing anything in this setup?? it used to work .. and it STILL works on another hosted server with Agents located in Morocco.. with a different version of Asterisk 1.4.20-1 and better hold time for the calls.. -- AHD Tarek Sawah _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Case.
Greetings again List. I'm facing a strange case with one of the productive Asterisk servers.. i have 3 providers sending traffic to the call center where agents pickup the calls. calls come into the server Queue Agents Last October .. an undersea cable got disconnected placing Egypt and the countries in the region offline.. when internet came back .. the call center located in Egypt had no SIP protocol working.. and we shifted to IAX.. 26 days later SIP started to work again .. but since then calls started to disconnect out of the blue.. we get calls that may last for 45 minutes.. and end normaly .. and we get calls that ring and disconnect the moment the agent picks up been facing a problem with my client as they use the Flash Operator Panel to monitor the call flow through the server and the regualr setup Queue Local users won't work for them as the Flash operator flash offline static agents as online so the client won't know who is on and who is off.. and it's impossible to teach the agents to Login and Logoff the Queue.. so the only solution is the following.. Caller Queue FindMeFollowMe Extension Local SIP extensions this way .. my client is able to monitor the calls and things won't get complicated.. (this is the setup we have been using for 6 months before the problem with the internet occures) since the internet problem and calls are getting disconnected .. out of the blue.. nothing has changed.. and to make sure things are going well .. we moved the server to a Hosting company in California with 10 mb/s connection speed.. (Same Setup that was working well) and still calls get disconnected.. after a lot of problems with the client .. i asked them to change the ISP (my prime suspect was the internet) and finaly they managed to change the ISP .. but the problem is still there.. my server informations are the following Asterisk 1.4.22-3 Uname -a: Linux 2.6.18-92.1.18.el5 sip.conf ;;Agent Sample from Sip.conf [3000] type=friend secret=3000 qualify=yes port=5060 disallow=all allow=g729 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/3000 context=from-internal canreinvite=no call-limit=1 busy-limit=1 ;;Provider's Sample from Sip.conf [50011] type=peer qualify=yes port=5060 pickupgroup= nat=no host=XXX.YYY.ZZZ.NNN disallow=all allaw=alaw allaw=ulaw allow=g729 dial=SIP/50011 context=from-internal canreinvite=no deny=0.0.0.0/0.0.0.0 permit=XXX.YYY.ZZZ.NNN/255.255.255.255 # extensions.conf ;;the provider sends calls to Virtual DIDs (Extensions) in my system which is 8000 exten = 8000,1,GotoIfTime(07:00-16:00|sat-fri|1-31|jan-dec?ext-queues,*8000,1) exten = 8000,n,Answer exten = 8000,n,Queue(8000,t,,,10) exten = 8000,n,Dial(IAX2/6005:6...@backupserver/11) ;; sends the call to a backup server. exten = *8000,1,Answer exten = *8000,n,Dial(IAX2/6005:6...@backupserver/11) the Providers strictly send calls with codec G.729 my agents get best voice quality with G.711u I need your advice .. am i missing anything in this setup?? it used to work .. and it STILL works on another hosted server with Agents located in Morocco.. with a different version of Asterisk 1.4.20-1 and better hold time for the calls.. -- AHD Tarek Sawah _ Get your vacation photos on your phone! http://windowsliveformobile.com/en-us/photos/default.aspx?OCID=0809TL-HM___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Vyatta routers solving NAT problems
Greetings again list i've seen plenty of posts talking about Asterisk behind nat .. and i was wondering.. have you ever thought of using Vyatta? i've been using it for more than two years.. and i'm sure it's a great addition to the open source community .. i DID install Asterisk behind vyatta and configured the nat .. system up and running smoothly .. if anyone else have tried it please let me know if any problems have been faced Regards -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Get free photo software from Windows Live http://www.windowslive.com/online/photos?ocid=PID23393::T:WLMTAGL:ON:WL:en-US:SI_PH_software:082009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Vyatta routers solving NAT problems
First of all it acts like a firewall and a router.. compared to Cisco routers it has good ACL and firewall policies that can be used and written very well.. second it's easy to setup third my question is has anyone tested it ? and what are their openion regarding this? if the question is not supposed to be directed to this list then just disregard it my friend -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Tue, 4 Aug 2009 07:32:15 -0400 From: abalas...@evaristesys.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Vyatta routers solving NAT problems Tarek Sawah wrote: Greetings again list i've seen plenty of posts talking about Asterisk behind nat .. and i was wondering.. have you ever thought of using Vyatta? i've been using it for more than two years.. and i'm sure it's a great addition to the open source community .. i DID install Asterisk behind vyatta and configured the nat .. system up and running smoothly .. if anyone else have tried it please let me know if any problems have been faced I am not sure that I understand your question. What precisely is the problem which Vyatta is intended to solve here, and why does it solve it better? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get free photo software from Windows Live http://www.windowslive.com/online/photos?ocid=PID23393::T:WLMTAGL:ON:WL:en-US:SI_PH_software:082009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and E1 Cards
Greetings List, i have a new question regarding Asterisk and E1 Cards a client of mine is requiring an Asterisk Server with 2 E1s. the scenario is the following they want 400 extensions to register with the system.. and required 64 concurrent calls. added to it that they are expecting the system to have an IVR to do some DB querying. the setup I have in mind is a Core2duo Server with 3 GB Ram and a Raid0 and a TE220B card. we have not faced this need from a client as we usually provide SIP Services only.. so my questions are the following 1- how many calls my setup will be able to handle? and if it won't handle 2 E1s what is the best server i can get for that? 2- E1 supports Ulaw and Alaw codecs so we won't be needing G729 nor G723 encoding and decoding? or we will have to use such codecs? (I'm concirned about the System resources) Thank you in Advance for your help and support. regards Tarek _ Get free photo software from Windows Live http://www.windowslive.com/online/photos?ocid=PID23393::T:WLMTAGL:ON:WL:en-US:SI_PH_software:082009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and E1 Cards
do you suggest buying a licensed Software from Digium? Date: Sun, 2 Aug 2009 18:53:16 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and E1 Cards On Sun, Aug 2, 2009 at 6:37 PM, Tarek Sawah tareksa...@hotmail.com wrote: Greetings List, Greetings i have a new question regarding Asterisk and E1 Cards a client of mine is requiring an Asterisk Server with 2 E1s. the scenario is the following they want 400 extensions to register with the system.. and required 64 concurrent calls. Unless I am mistaking, or you are including internal calls, 2 E1 would handle 62 or 60 if PRI. 400 extensions should be no problem. On the same LAN? added to it that they are expecting the system to have an IVR to do some DB querying. the setup I have in mind is a Core2duo Server with 3 GB Ram and a Raid0 and a TE220B card. Hard to say which would be better, two lower spec (cheaper) boxen setup identically, one as a cold swap. Backup DB, conf, and whatever, nightly. I have done this for many customers. RAID 0 is basically useless for Asterisk and sets yourself up for double chance of disk failure. RAID 1 is the way to go. we have not faced this need from a client as we usually provide SIP Services only.. so my questions are the following 1- how many calls my setup will be able to handle? and if it won't handle 2 E1s what is the best server i can get for that? You can handle that easily unless you are doing heavy codecs like G729 or recording every call. 2- E1 supports Ulaw and Alaw codecs so we won't be needing G729 nor G723 encoding and decoding? or we will have to use such codecs? (I'm concirned about the System resources) You should have said that first ;) A pentium 4 2.8ghz could handle this without breaking a sweat. Thank you in Advance for your help and support. regards Tarek -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _ Get free photo software from Windows Live http://www.windowslive.com/online/photos?ocid=PID23393::T:WLMTAGL:ON:WL:en-US:SI_PH_software:082009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] regarding to field of accountcode
accountcode is a setting you add to your SIP peer.. so it doesn't require restarting Asterisk.. only restart the SIP module.. sip reload will be enough my friend.. -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Fri, 29 May 2009 17:21:08 +0800 From: maillist...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] regarding to field of accountcode Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? 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 _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_BR_life_in_synch_052009___ -- 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] SIP Trunk groups
i'm not so familiar with what youa re talking about .. but i beleive i've seen
something like that in FreePBX where you can setup a failover trunk for a
context.. try to have a look at it. and i hope it's what you are looking for
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
From: mlecu...@gmail.com
Date: Wed, 27 May 2009 14:17:23 -0300
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Trunk groups
Hey all,
I have 2 GSM to Voip gateways and probably we will grow
up to 4 more gateways. I already created a macro to make failover
happen between gateways, but can imagine that everytime I add a new
gateway I will need to modify the macro. The initial intention of this
macro was to failover between different techonolgies.
So I was hoping to create a Sip Trunk group using the same idea as truckgroup
under dahdi but for sip trunks.
Is that possible?, have you ever done this before?
My Idea is:
sip_trunk1 = SIP/gateway1
sip_trunk2 = SIP/gateway2
sip_trunk3 = SIP/gateway3
gsm_trunkgoup = sip_trunk1 ; sip_trunk2 ; sip_trunk3
[user]
exten = _0.,1,wait()
exten = _0.,n,Dial(gsm_trunkgoup/${
exten:1},30)
exten = _0.,n,Hangup
Thanks,
--
--
Mariano Lecuona
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Re: [asterisk-users] connection fail between Service provider's proxy server and my asterisk server
some how the extension you have identified in your extensions.conf file is
wrong..
you are forwarding your call to an extension @ a local extension??
you can try at least the following
[default]
exten = _X.,1,Dial(SIP/${ext...@proxy.sp.co.kr)
it may work .
let me know
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Fri, 29 May 2009 12:05:36 +0200
From: megaho...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] connection fail between Service provider's
proxy server and my asterisk server
2009/5/29 김무성 ki...@infosec.co.kr
I wanna connect proxy server.
my IP Phone - my asterisk - service
provider's proxy server - extern PSTN phone
but asterisk server can't register to proxy
server.
I think that configuration is right.
When asterisk send to register request,
proxy server don't response.
I did capture packet. but no response.
MY setting
sip.conf
[kms]
username=kms
type=friend
secret=
host=dynamic
nat=yes
qualify=yes
callerid=0134
register = 0700134:passw...@proxy.sp.co.kr:5060/0134
[my-out]
type=peer
host=SP's proxy IP
username=0700134
secret=password
fromuser=0700134
fromdomain=proxy.SP.co.kr
extensions.conf
[default]
exten = _X.,1,Dial(SIP/${ext...@my-out)
If lines provided is not a form of trunk,
can't my asterisk server connect to proxy?
I could connect my IPPhone to proxy
directly.
but asterisk not.
We need the sip trace for the call.
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Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
through a test .. i was able to send calls from Asterisk 1.4 to a PSTN number
through a cisco router with a channel bank.. Audio worked well.. i setup a
dial plan in asterisk to Dial(${ext...@ciscoip) and authorise the cisco
router's ip on the asterisk server and treat the calls comming from it like any
other SIP calls inside the server..
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Sat, 16 May 2009 14:46:27 +0300
From: timotsm...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Hi,
In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company (which will be the same as phone DID), Voice
mail, Auto Answer etc yet we need thousands of dollars to add those to
our cisco call manager 4.1 set up.
I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to change codec but no sucess!
Has anyone had the above set up working successfully? Attached are some confs.
Thanks a lot for your assistance.
Kind Regards,
Wilson
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Re: [asterisk-users] no source on calllogs
try adding callerid=CIDNAME CIDNUM this will force your callerID in your DIalplan -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Wed, 29 Apr 2009 09:38:58 +0300 From: oguzh...@bilkent.edu.tr To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] no source on calllogs just post your peer configs for one of your clients that don't show on the log. mostly it's IAX peers that don't show on the logs if not configured to. All my clients are sip peers actually. Here is the users.conf entry for one of the users that doesnt show on logs. [8006] username = 8006 transfer = yes mailbox = 8006 call-limit = 100 fullname = Test registersip = no host = dynamic callgroup = 1 call-limit = 100 context = DLPN_All cid_number = 8006 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = yes callwaiting = yes hasmanager = no hasagent = no hassip = yes hasiax = yes secret = nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 autoprov = no label = macaddress = linenumber = 1 LINEKEYS = 1 -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Tue, 28 Apr 2009 13:15:12 +0300 From: oguzh...@bilkent.edu.tr To: asterisk-users@lists.digium.com Subject: [asterisk-users] no source on calllogs Hello, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using Asterisk 1.6.0.9 (had the same problem with 1.6.0.6 too) 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 _ Windows Live™ Hotmail®:…more than just e-mail. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_more_042009___ -- 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 _ Windows Live™ Hotmail®:…more than just e-mail. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_more_042009___ -- 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] Video Conference Software (Open Source)
from my expreience .. if you don't setup a CALLER ID in your PEER that your second PBX is registering with .. it will pass any caller ID in the header give it a try .. Salam! -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Mon, 27 Apr 2009 09:57:14 +0700 From: joko.pit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Video Conference Software (Open Source) I am looking for Video Conference Software (Open Source) , But but not for free Trial.. please give reference about it. Thanks _ Windows Live™ SkyDrive™: Get 25 GB of free online storage. http://windowslive.com/online/skydrive?ocid=TXT_TAGLM_WL_skydrive_042009___ -- 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] no source on calllogs
just post your peer configs for one of your clients that don't show on the log. mostly it's IAX peers that don't show on the logs if not configured to. -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Tue, 28 Apr 2009 13:15:12 +0300 From: oguzh...@bilkent.edu.tr To: asterisk-users@lists.digium.com Subject: [asterisk-users] no source on calllogs Hello, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using Asterisk 1.6.0.9 (had the same problem with 1.6.0.6 too) 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 _ Windows Live™ Hotmail®:…more than just e-mail. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_more_042009___ -- 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] Serving 120 concurrent calls
Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. we are planning on using 4 servers to serve the calls and one for the database servers' specifications are: 2.8 Dual Core Proccessors 2 GB Ram 160 Sata Drive each server will be provided with 1 E1 card Questions are: 1- will those servers be able to handle that ammount of calls?' 2- the important issue is that they require call recording on all calls.. which means we will have to record ALL calls going out of the system .. which means we will need a call recroding.. will the four Asterisk servers handle the recording process or we will need external assistant? and if it was the second choice what is the best suggestion? is there a way to force an Asterisk server to record remote channels? -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Express your personality in color! Preview and select themes for Hotmail®. http://www.windowslive-hotmail.com/LearnMore/personalize.aspx?ocid=TXT_MSGTX_WL_HM_express_032009#colortheme___ -- 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] DID provider in Sweden
try the following http://www.callcentric.com they are the best i've ever dealt with .. they provide did numbers in Sweden-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Wed, 10 Dec 2008 15:30:59 + From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DID provider in Sweden On Wed, 10 Dec 2008, Peter Lindquist wrote: Hi Gordon, Take a look at http://www.cellip.com/ Ah! Thanks! I'll pass it on. Gordon//Peter Gordon Henderson wrote: On Wed, 10 Dec 2008, Gideon Hack wrote:Hi Gordon, DID World Wide (see http://www.didww.com/virtual_numbers/Sweden) has the DIDs that you require. And they can forward to IAX if that is preferable to you.Thanks. I was actually hoping I'd find a Swedish company, but I'll pass this and the other on to my customer (who's in Sweden and wants to pay in Swedish money) Cheers, Gordon ___ -- 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 _ Send e-mail anywhere. No map, no compass. http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_anywhere_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] Func_ODBC question
if you are using MYSQL.. why don't you query your DB directly from Asterisk ?
the following example is something i use with my servers
[ivr1-cont]exten = 7700,1,Answer
exten = 7700,n,MYSQL(Connect connid 127.0.0.1 root rootpass TarekDB)exten =
7700,n,MYSQL(Query resultid_2 ${connid} SELECT q_name FROM tbl_ivr ORDER BY
RAND( ) LIMIT 1 )exten = 7700,n,MYSQL(Fetch fetchid1 ${resultid_2}
question)exten = 7700,n,Read(A1,ivr1/${question})exten =
7700,n,MYSQL(Disconnect ${connid})
-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria:
+963 944 618286 USA: +1 347 562 2308
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 9 Dec 2008 15:45:59
-0200Subject: [asterisk-users] Func_ODBC question
Hi I have
On func_odbc
[EXEC]
readhandle=ressqlserver
writehandle=ressqlserver
readsql=${ARG1}
writesql=${ARG1}
I’m trying an update on dialplan:
exten= 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})})
On Cli:
WARNING[3579]: func_odbc.c:353 acf_odbc_read: Error -1 in FETCH [UPDATE Tabla
set campo = 4356]
Any idea why is this??
The query works fine, I just wanto to know if the warning can cause any problem
to me.
Thanks!!
Sebastian
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