[Asterisk-Users] RE: AS5300+SIP+ASTERISK or AS5300+MGCP
AS5300 setup =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2005.04.04 09:37:31 =~=~=~=~=~=~=~=~=~=~=~= sh runn Building configuration... Current configuration : 11599 bytes ! ! Last configuration change at 03:26:25 GMT Mon Apr 4 2005 by charles ! NVRAM config last updated at 03:06:50 GMT Mon Apr 4 2005 by charles ! version 12.3 service timestamps debug datetime localtime show-timezone service timestamps log datetime localtime show-timezone service password-encryption ! hostname 66.178.100.66 ! boot-start-marker boot-end-marker ! ! ! resource-pool enable --More-- ! resource-pool throttle 20 default clock timezone GMT 3 clock calendar-valid ! aaa new-model ! ! aaa group server radius cdt-1 server 159.148.8.108 auth-port 2362 acct-port 2363 ! aaa group server radius cdt-2 server 62.85.77.82 auth-port 2362 acct-port 2363 ! aaa group server radius tsl server 62.56.250.200 auth-port 1812 acct-port 1813 ! aaa authentication login h323 group radius aaa accounting send stop-record authentication failure aaa accounting connection h323 stop-only broadcast group cdt-1 group cdt-2 group tsl aaa nas port voip aaa session-id common --More-- ip subnet-zero ip telnet source-interface FastEthernet0 ip name-server 66.178.100.68 ! ! ! trunk group mgcp ! isdn switch-type primary-net5 ! voice rtp send-recv ! voice service pots ! voice service voip cause-code legacy h323 h225 timeout setup 8 session transport udp sip min-se 600 ! voice class codec 2 --More-- codec preference 1 gsmfr ! ! voice class permanent 1 signal timing idle suppress-voice 5 signal timing oos suppress-all 30 signal timing oos timeout 120 ! ! voice class h323 1 h225 timeout tcp establish 30 h225 timeout connect 60 h225 timeout setup 30 call start fast ! voice class h323 2 call start slow ! voice class h323 1001 call start fast ! voice class h323 10 ! --More-- ! voice class busyout 1 ! ! voice class dualtone-detect-params 1 ! ! ! ! ! fax interface-type modem ! ! controller E1 0 clock source line primary ds0-group 0 timeslots 1-15,17-31 type r2-digital ! ! ! translation-rule 22 Rule 0 22254 254 ! ! ! interface Tunnel1 ip address 192.168.44.1 255.255.255.0 tunnel source Ethernet0 tunnel destination 217.21.95.9 ! interface Tunnel17 --More-- ip address 10.1.17.2 255.255.255.0 shutdown tunnel source 212.165.147.254 tunnel destination 66.92.133.199 tunnel mode nos ! interface Tunnel18 no ip address ! interface Ethernet0 ip address 195.202.73.106 255.255.255.248 no ip mroute-cache ! interface Serial0 no ip address no ip mroute-cache clockrate 2015232 no fair-queue ! interface Serial1 no ip address no ip mroute-cache clockrate 2015232 --More-- no fair-queue ! interface Serial2 no ip address no ip mroute-cache clockrate 2015232 no fair-queue ! interface Serial3 no ip address no ip mroute-cache shutdown clockrate 2015232 fair-queue 100 256 0 ip rtp priority 1 1 75 ! interface Serial2:15 no ip address isdn switch-type primary-net5 no cdp enable ! interface FastEthernet0 ip address 172.16.202.90 255.255.255.0 secondary --More-- ip address 66.178.100.66 255.255.255.248 ip access-group 1 in ip access-group 1 out no ip mroute-cache duplex auto speed auto h323-gateway voip interface h323-gateway voip id gk0 ipaddr 216.52.153.203 1719 h323-gateway voip h323-id ngins ip rtp priority 16384 16383 400 ! ip classless ip route 0.0.0.0 0.0.0.0 66.178.100.65 no ip http server ! ! no logging trap access-list 101 permit ip any any ! route-map VOIP permit 20 match ip address 101 ! route-map VOIP permit 100 --More-- ! ! radius-server attribute 44 include-in-access-req radius-server host 159.148.8.108 auth-port 2362 acct-port 2363 key 7 065E582A585C51411F0317 radius-server host 62.85.77.82 auth-port 2362 acct-port 2363 key 7 014B510F4F195E573B584B radius-server host 62.56.250.200 auth-port 1812 acct-port 1813 key 7 121500031F0E050A radius-server retransmit 10 radius-server timeout 120 radius-server vsa send accounting radius-server vsa send authentication call threshold global total-calls low 60 high 90 busyout ! call application voice kenya flash:kenya.tcl ! call application voice kenya1 flash:kenya.tcl ! ! voice-port 0:0 compand-type a-law connection plar 9001 ! ! mgcp call-agent 62.56.250.198 2427 service-type mgcp version 1.0 mgcp dtmf-relay voip codec all mode out-of-band mgcp restart-delay 2 mgcp codec g711ulaw packetization-period 10 mgcp package-capability dtmf-package mgcp package-capability line-package mgcp package-capability rtp-package mgcp package-capability nas-package mgcp package-capability script-package mgcp sdp simple --More-- no mgcp validate domain-name mgcp endpoint offset mgcp bind control source-interface FastEthernet0 mgcp bind media source-interface FastEthernet0 mgcp behavior signals v0.1 ! mgcp profile default ! dial-peer cor custom ! ! ! ! dial-peer voice 271 pots permission orig huntstop application
[Asterisk-Users] Previous sip reload not yet done
Hi list, We are running a CVS version of 03-30-2005 but also had this behaviour on previous versions. From time to time, after a period of not making calls (eg a night or few hours), we have no dialtone when we want to call. "SIP show peers" show EP registered with status OK but nothing happend. Nothing special in logs. After a "SIP reload", everything is again working fine. So we add a "SIP reload" each morning in crontab. But this is not solving the problem: it's not always efficient and when we try to re-run this command from CLI, we get a "Previous sip reload not yet done". Only solution is to restart asterisk. Does anyone else have this problem? Is there a workaround? Thanks for any hints. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Realtime Capabilities
The term RTCache has never been mentioned in the WIKI or these forums. I assume that it's some sort of function to speed up realtime db access by keeping transactions in RAM and writing periodically? If so, I can understand why this would need to be flushed. - Original Message - From: "Matthew Boehm" <[EMAIL PROTECTED]> To: "Asterisk Users" Sent: Monday, April 04, 2005 3:01 PM Subject: Re: [Asterisk-Users] Asterisk Realtime Capabilities to a load-balanced (not sure which mechanism I'll empoy here yet) I was quoted a $21,000 layer-7 switch from F5 Networks to do SIP load balancing. outside)? In other words, can the registering server update a USRLOC type database on the fly, so all other servers know where to route calls As long as all * servers share the same central database; this way when SIP 1 registers via RealTime at server A, server B (using same db) should be able to see the registration. You may not be able to use RTCache though... Also, I will be using multiple * boxes as media gateways. Is there an existing mechanism whereby a given server can record the number of busy/available ZAP channels to a central database for the purpose of call routing? Nothing built-in comes to mind, but Im sure you could AGI something. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to configure groups using a sip phone
Can you be more specific? What are you trying to achieve with the creation of such "groups"? - Original Message - From: <[EMAIL PROTECTED]> To: Sent: Monday, April 04, 2005 3:50 PM Subject: [Asterisk-Users] how to configure groups using a sip phone hi friends ! i am facing a problem from one week and now required ur help urgently. Actually, i want to configure asterisk for two groups javgroup and linuxgroup. i also have constraint to use only sip phone (esatara ). now, please help me is it possible to configure astersik in that way or that kind of facility is given in zapata.conf. tell me in detail abt the configurations of the sip.conf and extensions.conf. thanks Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to configure groups using a sip phone
hi friends ! i am facing a problem from one week and now required ur help urgently. Actually, i want to configure asterisk for two groups javgroup and linuxgroup. i also have constraint to use only sip phone (esatara ). now, please help me is it possible to configure astersik in that way or that kind of facility is given in zapata.conf. tell me in detail abt the configurations of the sip.conf and extensions.conf. thanks Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression
Alejandro G wrote: I have a problem with ATA-186 configured for silence supression (AudioMode bit 0 = 1). When enabled and listening music on hold no sound is heared (if I talk I began to hear the music and again mutes when I stop talking). If I configure for silence supression off everything goes fine. Any hint? Don't use silence supression. Asterisk doesn't support it. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] V92 modem with asterisk
Hi everyone, I just install Linux and asterisk on one of my pc. I want to run some basic functionality tests. Is it possible to use a v92 modem as a FXO or FXS card. If yes how do we configure and install the card? What are the commands? Thanks in advance for your help AC __ Lèche-vitrine ou lèche-écran ? magasinage.yahoo.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: X100P interrupt load
On 03/23/05 04:15 Jesse Guardiani said the following: This should be "has some issues". I do not consider the FreeBSD zaptel support to be production quality in any way. I experienced reproducible system hangs (mostly after an asterisk restart), interrupt issues (audio skips and SSH pauses during typing), and general instability. This was with an up-to-date FreeBSD 5.3-SECURITY and the latest zaptel at asterisk from ports (1.0.6 for asterisk, and a significantly lower version for zaptel, I think). I do not recommend anyone run FreeBSD + Asterisk at this time. perhaps a post detailing how these hangs happenned and any CLI output before these hangs would help in /eliminating/ this. i'm running asterisk on freebsd 4.x /with/ digium TDM cards without any problems. any problems i faced were usually tied down the the digium hardware itself, instead of asterisk or freebsd. note that "noload => pbx_wilcalu" needs to exist in modules.conf, as detailed in the asterisk on freebsd wiki. not a hardware guy, so I don't know much about interrupts. Just that 1000 interrupts/sec is fairly high. :) those are the interrupts which the digium cards generate, and are used for timing. it's not specifically a freebsd issue. -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music On Hold and ATA-186 w/Silence Supression
Hi, I have a problem with ATA-186 configured for silence supression (AudioMode bit 0 = 1). When enabled and listening music on hold no sound is heared (if I talk I began to hear the music and again mutes when I stop talking). If I configure for silence supression off everything goes fine. Any hint? Anybody with same problem? Thanks. Alejandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home Question
> I was wondering if there is a way to select the outbound > trunk based on the extension that making the call. Set the context in the sip.conf file for that user to a context in extensions.conf that only has entries for dialing out through specific providers. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Realtime Capabilities
> to a load-balanced (not sure which mechanism I'll empoy here yet) I was quoted a $21,000 layer-7 switch from F5 Networks to do SIP load balancing. > outside)? In other words, can the registering server update a USRLOC > type database on the fly, so all other servers know where to route calls As long as all * servers share the same central database; this way when SIP 1 registers via RealTime at server A, server B (using same db) should be able to see the registration. You may not be able to use RTCache though... > Also, I will be using multiple * boxes as media gateways. Is there an > existing mechanism whereby a given server can record the number of > busy/available ZAP channels to a central database for the purpose of > call routing? Nothing built-in comes to mind, but Im sure you could AGI something. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home Question
Greetings! This is my first post to the list...and I'm kinda' new to Asterisk, so please be kindI did a fair amount of Googling but was not able to find an answer. I am using [EMAIL PROTECTED] 0.8 I was wondering if there is a way to select the outbound trunk based on the extension that making the call. Here is why I ask. Since I am already running my Asterisk server for my own use, I also wanted to let friends and family in on the action but I don't want to pay for their calls. So if I ask them to buy talk time from a termination provider and then setup a separate trunk for them, how do I make sure that only their calls use that outbound trunk? Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home Question
Greetings! This is my first post to the list...and I'm kinda' new to Asterisk, so please be kindI did a fair amount of Googling but was not able to find an answer. I am using [EMAIL PROTECTED] 0.8 I was wondering if there is a way to select the outbound trunk based on the extension that making the call. Here is why I ask. Since I am already running my Asterisk server for my own use, I also wanted to let friends and family in on the action but I don't want to pay for their calls. So if I ask them to buy talk time from a termination provider and then setup a separate trunk for them, how do I make sure that only their calls use that outbound trunk? Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI Dial Plan
Hi everyone
Presently all our calls are channel to one provider and we would like to change that based on LCR.
the following is what we have presently;
# Dial the requested number, if we got something from the caller.if ($dialto != ""){ $AGI->exec('SetAccount', $accountnum); if ($debug) { $AGI->exec('NoOp', "\"Dialing $dialto... \""); } $AGI->exec('Dial', "Zap/g2/$dialto|30|C");}$AGI->hangup();
How do i make AGI dial to g1 in the event that user enter area code 416 ?
any assistant is greatly appreciated
regards
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[Asterisk-Users] creating conference call
Hi, I am looking at a project using asterisk for a particular purpose. We already are using an Asterisk box for things like voicemail, call recording, ip phones etc. and it connects to an old standard PBX through ZAP. What I am looking to do is have calls coming into asterisk via either VOIP or ZAP which are then connected through ZAP to the old PBX and out into the normal PSTN. That part I can handle, what I need is to be able to programatically put the external channel on hold, create a conference by dialling into another ZAP channel then after some conversation bringing back the one on hold into the conference, the the original call from the VOIP/ZAP can be hung up leaving the other two together. So basically is it possible with Asterisk to: - identify the channel by outgoing ph no. - put the channel on hold - dial through a new channel with conference - bring back the hold - drop the original channel and leave the others From what I understand of the way asterisk works this should be technically possible but I don't know of any way to implement this, maybe I need to make some change to the asterisk code? I cannot do the hold/conference from the phone or originating switch, this is a few interfaces down the line, I just want to be able to put asterisk in the middle to perform this function most likely through some computer program (I guess it would also be possible to detect a DTMF sequence to trigger the action). Does anyone have any ideas about how I could implement this? Any help is appreciated. Keiron Liddle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk <-> Altigen
> -Original Message- > [mailto:[EMAIL PROTECTED] On Behalf Of > Dan Perik > Subject: [Asterisk-Users] Asterisk <-> Altigen > > > Has anyone successfully tied together an Altigen system to an > Asterisk system using VoIP (ie. not using hardware (FXO/FXS > cards, etc.))? My experience with the Altigen's IP stack is a bit dated, so take this for what it's worth... At the time I was working with it, their VOIP implementation was so bad, that we abandoned it, and resorted to connecting spare analog ports to a Multi-Tech VOIP gateway. This solution worked like a champ. Even if Altigen's VOIP implemenation has gotten more solid, I'd recommend against using it, if for no other reason than the fact that it uses H.323. The H.323 support in Asterisk is spotty. In certain configurations, it seems to work fine, but others, H.323 <--> SIP, for example, it seems to "have issues". If you have much time to spare, and you already have the VOIP licenses for the Altigen, I guess you've got nothing to lose, but I wouldn't try it under any other terms. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 04/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk <-> Altigen
Hi, If this belongs on a different list, please let me know. I oversee an Altigen IP-based PBX. We're wanting to make VoIP calls through the Internet out to PSTN via a service like BroadVoice or similar. I think Asterisk is the ticket of this. I have successfully configured Asterisk to dialout/dialin on BroadVoice, FWD, etc. to/from X-Lite softphone. Altigen uses H.323, and can be configured for IP-based "trunk" access. Does anyone know if it would work to have the Altigen system "trunk out" calls via Asterisk as a gateway, and then Asterisk can connect them out via BroadVoice. Has anyone successfully tied together an Altigen system to an Asterisk system using VoIP (ie. not using hardware (FXO/FXS cards, etc.))? Thanks, Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP
If u want some help put your 53xx and sip config files.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jafar mohammed
Sent: Sunday, April 03, 2005 9:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP
hi's
i have been trying to configure my AS5300 to work with my asterisk box. i
have tried SIP, calls come, answered and AS5300 sends BYE message after not
more than 5 secs. I have also tried MGCP, but i believe i am not configuring
that right. here is the output of the sip debug. please help me out or lead
me to the direction of sorting this problem out.
thank you
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To:
Date: Mon, 04 Apr 2005 00:16:50 GMT
Call-ID:
[EMAIL PROTECTED]
Supported: timer
Min-SE: 600
Cisco-Guid:
2899651584-2748649945-2861211020-3122285050
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID:
;party=calling;screen=no;privacy=off
Timestamp: 1112573810
Contact:
Expires: 180
Allow-Events: telephone-event
MIME-Version: 1.0
Content-Type: multipart/mixed;boundary=uniqueBoundary
Content-Length: 431
--uniqueBoundary
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 5042 571 IN IP4
66.178.100.66
s=SIP Call
c=IN IP4 66.178.100.66
t=0 0
m=audio 18992 RTP/AVP 3 19
c=IN IP4 66.178.100.66
a=rtpmap:3 GSM/8000
a=rtpmap:19 CN/8000
a=ptime:10
--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional
IAM,
GCI,acd52c00a3d511d9aa8a9d8cba1a49fa
--uniqueBoundary--
-*-
- 21 headers, 21 lines
* Using latest SIP request as basis request
* Sending to 66.178.100.66 : 5060 (NAT)
Apr 4 00:16:40 NOTICE[25296]: chan_sip2.c:5872
check_user_full: User name from URI: 66.178.100.66, Digest auth user: (null)
== Authentication turned off, no secret for user
66.178.100.66
* No RDNIS header in SIP packet
-- - SIPFromURI:
;tag=8CB7504-1904
--> Transmitting (no NAT) response to
66.178.100.66:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Call-ID:
[EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
-*-
-- Executing Answer("SIP/66.178.100.66-bf34", "") in new stack
* SDP preparation: We're at 62.56.250.198 port 17962
* Answering with preferred capability 0x2 (gsm)
* Answering with preferred capability 0x4 (ulaw)
* Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1(g723)
--> Reliably Transmitting (no NAT) response to
66.178.100.66:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Call-ID:
[EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 25296 25296 IN IP4 62.56.250.198
s=session
c=IN IP4 62.56.250.198
t=0 0
m=audio 17962 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
-*-
-- Executing Wait("SIP/66.178.100.66-bf34", "2") in new stack
--- Sip read from 66.178.100.66:50341
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Date: Mon, 04 Apr 2005 00:16:50 GMT
Call-ID:
[EMAIL PROTECTED]
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK
-*-
- 9 headers, 0 lines
--- Sip read from 66.178.100.66:53065
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Date: Mon, 04 Apr 2005 00:16:50 GMT
Call-ID:
[EMAIL PROTECTED]
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 6
Timestamp: 1112573810
CSeq: 102 BYE
Content-Length: 0
-*-
- 11 headers, 0 lines
* Sending to 66.178.100.66 : 5060 (non-NAT)
--> Transmitting (no NAT) response to
66.178.100.66:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
-*-
== Spawn extension (AS5300, 9001, 2) exited non-zero on
'SIP/66.178.100.66-bf34'
-- Executing Hangup("SIP/66.178.100.66-bf34", "") in new stack
== Spawn extension (AS5300, h, 1) exited non-zero on
'SIP/66.178.100.66-bf34'
Destroying SIP dialogue '[EMAIL PROTECTED]'
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Asteris
Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
Rod Bacon wrote: This is quite interesting. I tested calls to 2 mobiles that I knew were off, and not diverted to voicemail. 1 with Telstra, the other with vodafone (I'm in Australia). Via ISDN, both calls were shown as unanswered by asterisk. When the calls went to voicemail, the call was deemed to be answered. Via analogue circuits, the call is shown as answered, no matter what. That's what I would expect. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 21
hello can any one tell me what is the problem in my odbc connection. here is my sql.log connection with mysql is working and with freetds is giving me error jawad is one windows server having MS Sql server #isql kdsn src/tds/login.c: tds_connect: jawad:1433: Connection refused [ISQL]ERROR: Could not SQLConnect [ODBC][3085][__handles.c][444] Exit:[SQL_SUCCESS] Environment = 0x823cb58 [ODBC][3085][SQLAllocHandle.c][345] Entry: Handle Type = 2 Input Handle = 0x823cb58 [ODBC][3085][SQLAllocHandle.c][463] Exit:[SQL_SUCCESS] Output Handle = 0x823d130 [ODBC][3085][SQLConnect.c][3526] Entry: Connection = 0x823d130 Server Name = [kdsn][length = 7 (SQL_NTS)] User Name = [kami][length = 11 (SQL_NTS)] Authentication = [***][length = 11 (SQL_NTS)] UNICODE Using encoding ASCII 'ISO8859-1' and UNICODE 'UCS-2LE' DIAG [08S01] [FreeTDS][SQL Server]Server is unavailable or does not exist. DIAG [S1000] [FreeTDS][SQL Server]Unable to connect to data source [ODBC][3085][SQLConnect.c][3894] Exit:[SQL_ERROR] [ODBC][3085][SQLFreeHandle.c][268] Entry: Handle Type = 2 Input Handle = 0x823d130 [ODBC][3085][SQLFreeHandle.c][317] Exit:[SQL_SUCCESS] [ODBC][3085][SQLFreeHandle.c][203] Entry: Handle Type = 1 Input Handle = 0x823cb58 /usr/local/etc/odbc.ini [mydsn] Description = test Driver = test Server = localhost Database= asterisk Port= 3306 Socket = Option = Stmt= [kdsn] Description = tds dsn Driver = tdsdriver Servername = Server = jawad Address = jawad Port= 1433 Database= kdb TDS_Version = 4.2 Language= us_english TextSize= Domain = PacketSize = /usr/local/etc/odbcinst.ini [ODBC] Trace = Yes TraceFile = /tmp/sql.log ForceTrace = Yes Pooling = No [test] Description = test Driver = /usr/lib/libmyodbc.so Driver64= Setup = /usr/local/lib/libodbcmyS.so.1 Setup64 = UsageCount = 1 CPTimeout = CPReuse = [tdsdriver] Description = driver for tds Driver = /usr/local/lib/libtdsodbc.so Driver64= /usr/local/lib Setup = /usr/local/lib/libtdsodbc.so.0 Setup64 = /usr/local/lib UsageCount = 1 CPTimeout = CPReuse = __ Do you Yahoo!? Yahoo! Personals - Better first dates. More second dates. http://personals.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Joshua Chessman
Empty yer bloody mailbox... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
This is quite interesting. I tested calls to 2 mobiles that I knew were off, and not diverted to voicemail. 1 with Telstra, the other with vodafone (I'm in Australia). Via ISDN, both calls were shown as unanswered by asterisk. When the calls went to voicemail, the call was deemed to be answered. Via analogue circuits, the call is shown as answered, no matter what. - Original Message - From: "Ian Hailey" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, April 04, 2005 5:56 AM Subject: [Asterisk-Users] Detecting when a called mobile is not reachable? Hello all, I was hoping to be able to call a mobile and if it is un-reachable for whatever reason (e.g. switched off) then I was expecting an unobtainable response that would be detected in Asterisk. It seems that the operator (Virgin in UK) imedately completes the call and plays an automated message before clearing the call. Does anyone know if there a way of avoiding the call completion for mobiles? I have noticed that Sipgate charge for a calls to an unavailable mobile regardless. Thanks. Ian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SET & CHECK group
Mark Halverson wrote:
exten => _1NXXNXX,1,SetGroup(${CALLERIDNUM})
Try using ${ACCOUNTCODE} and make sure the account code is unique to
each phone.
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Re: [Asterisk-Users] How to reset IAXy?
I put the Who? in Mishehu wrote: I'd put the device and another machine on a separate physical network where you can make whatever IP configurations you need in order to be able to send data to the IAXy. Then you can load new configuration to it there. There might be a better way to do i, but I don't know for sure. -mishehu Lam H. Nguyen wrote: Can anyone tell me how to reset the IAXy? I used I put it the wrong ip config in the IAXy and it conflicts with my network whenever I plug it in. Currently the DHCP is disable. I need to re-enable it to change the settings. The hard reset button on the IAXy doesn't seem to work The reset button works! You just need to keep pressing it while you turn on the power, and keep holding it for at least 4 hours (I know I am three days too late for that joke, ... ) bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
On Apr 3, 2005 8:56 PM, Ian Hailey <[EMAIL PROTECTED]> wrote: Hello all, I was hoping to be able to call a mobile and if it is un-reachable for whatever reason (e.g. switched off) then I was expecting an unobtainable response that would be detected in Asterisk. It seems that the operator (Virgin in UK) imedately completes the call and plays an automated message before clearing the call. Does anyone know if there a way of avoiding the call completion for mobiles? I have noticed that Sipgate charge for a calls to an unavailable mobile regardless. Bellsouth at least WILL play an automated message, but NOT answer the line. I work around this by adding the "r" option to the Dial command. The "r" option of course provides a fake ringing sound to the caller, even if it REALLY should be doing something else like playing telco audio before answer, or a busy tone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: using unixODBC
hello i dont know why unixodbc is not working. i am trying to make odbc connection. yesterday my odbc connection was working with mysql on my one mechine but now it is not working. is there any problem in code. /etc/odbc.ini [test] Description = My test dsn Trace = Off TraceFile = stderr Driver = mytestdriver SERVER = 127.0.0.1 USER = asteriskuser PASSWORD = asteriskpassword PORT = 3306 DATABASE = asteriskcdrdb /etc/odbcinst.ini [mytestdriver] Description = MySql driver for linux Driver = /usr/lib/libodbc-2.50.39.so __ Yahoo! Messenger Show us what our next emoticon should look like. Join the fun. http://www.advision.webevents.yahoo.com/emoticontest ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk
I think this problem is exactly the one I am having. The issue is: http://www.pastebin.com/266724 042 Found no matching peer or user for '192.168.17.1:56730' to which asterisk generates a "SIP/2.0 404 Not Found" (line 057) yet you have it configured here: [operator] type=peer canreinvite=no host=192.168.17.1 context=cme-pbx Hopefully someone with a working configuration can provide feedack. Regards, Nathan. Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks I've a strange problem, probably a mistake but I don't see it :( Description: My ephone-dn number on ccme, that is a simple connection plar for all ISDN calls, is 601 The voicemailmain on asterisk is 5900. CCME: 192.168.17.1 *: 192.168.17.10 My sip.conf: http://www.pastebin.com/266718 My extension.conf: http://www.pastebin.com/266720 My voicemail.conf: http://www.pastebin.com/266722 when I call the asterisk server from SIP free accounts, I receive the call on 601 (my 7960 phone) and then the call will be forwarded to voicemail without any problem. But when I receive a call from ISDN cloud, the 601 rings, the call is forwarded (see debug) on voicemail (number 5601), but the line goes down. This is the debug, that is I suppose the problem is on my Asterisk config (the 'ext-number' is the caller ID): http://www.pastebin.com/266724 I hope you could help me :) Thanks for all Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCUIYAMakHrsrHP9wRAlR1AKDKNzARotrmFMPphvjwqjp8da4SwACfQ6lo hxesZUu9t220j8zfQHW2DX0= =zJCw -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does asterisk know the did called on?
Courtney Couch wrote:
If I were to buy 20 did's how do I know within asterisk which number was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
optimum).
Your carrier can tell you how many digits they will send to you.
Asterisk sees these digits and will match exten => 1234,1,Blah if the
carrier sends you 4 digits.
Remember Asterisk does not really support DID on analog ports, only
T-1/E-1 (including PRI) ports , BRI ports, and VoIP ports.
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Re: [Asterisk-Users] Buying some Polycom IP300s
erm, how much u willing to sell ip500?, i would like to get 1 or 2 for my developments testing purposes. BTW if u do sell me, I'm in Malaysia, is it a problem for u to send it over? :D thanz. At 04:39 AM 4/4/2005, you wrote: We have a majority of IP300's, and a few IP500's. The IP300's are great phones if you need to simply drop in a bunch of VoIP phones quickly and cheaply. The IP300's simply lack certain features like speakerphone that you may want. Aside from that, its a great phone. -Courtney Dan Morin wrote: Sorry for the double post, I tried to paste and accidently sent the email I've been playing with Asterisk for a few weeks now, and I've gotten everything to work well with softphones, so I'm ready to move on to normal VoIP phones. I've been looking around and reading comments that people have had, and I was convinced that the Polycom IP300 was a great phone for a good price. But, then I ran into this page, which has been update in the last few days: http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500 The page in the wiki used to say that the person would not recomed Polycom phones to anyone. So anyway, I just want to make sure that the IP300 is a good choice. I don't want to get cheap phones that aren't business quality, since I do play on using them for my business after testing. Also, is the IP500 worth the extra money? What can it do that the IP300 can't. And finally, will the IP300 do ulaw encoding? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Realtime Capabilities
Hello all. I am trying to architect a large-scale solution and need to know some of the capabilities of * using realtime configuration (I have read some docuemntation on the WIKI, but have not yet played with Realtime). As the supporting docco is a little light-on at the moment, I'm hoping to get some meaningful feedback from those who've used it successfully. Essentially, I'm trying to build a distributed network of * boxes which will terminate SIP/IAX/PSTN calls. I would potentially like to use a central "farm" of redundant * boxes in place of a single SER box (if possible) to act as the SIP (and IAX) registrar/proxy. I know that SER is a much more efficient SIP proxy, but I'm hoping that the addition of multiple * boxes will largely negate this fact, and provide redundancy. Is it indeed conceivable to build a national network entirely of * servers, have users register via SIP/IAX to a load-balanced (not sure which mechanism I'll empoy here yet) farm of * servers (ie. an unknown server), and then receive calls from anywhere in the network (and outside)? In other words, can the registering server update a USRLOC type database on the fly, so all other servers know where to route calls for a given (dynamic?) client? Also, I will be using multiple * boxes as media gateways. Is there an existing mechanism whereby a given server can record the number of busy/available ZAP channels to a central database for the purpose of call routing? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
John Novack wrote: An even BETTER question is: When will what is already out and more or less working have enough accurate documentation to make it acceptable to a wider audience? Once more people start contributing. As one small example: the recent postings regarding wctdm. If all the options are at the end of the driver source, how long does it take to put into a more accessible form? Probably not long: just head over to http://asteriskdocs.org and see how you can help out, or pop on to the wiki ( http://www.voip-info.org ) and add an entry. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP
hi's
i have been trying to configure my AS5300 to work with
my asterisk box. i have tried SIP, calls come,
answered and AS5300 sends BYE message after not more
than 5 secs. I have also tried MGCP, but i believe i
am not configuring that right. here is the output of
the sip debug. please help me out or lead me to the
direction of sorting this problem out.
thank you
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To:
Date: Mon, 04 Apr 2005 00:16:50 GMT
Call-ID:
[EMAIL PROTECTED]
Supported: timer
Min-SE: 600
Cisco-Guid:
2899651584-2748649945-2861211020-3122285050
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID:
;party=calling;screen=no;privacy=off
Timestamp: 1112573810
Contact:
Expires: 180
Allow-Events: telephone-event
MIME-Version: 1.0
Content-Type: multipart/mixed;boundary=uniqueBoundary
Content-Length: 431
--uniqueBoundary
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 5042 571 IN IP4
66.178.100.66
s=SIP Call
c=IN IP4 66.178.100.66
t=0 0
m=audio 18992 RTP/AVP 3 19
c=IN IP4 66.178.100.66
a=rtpmap:3 GSM/8000
a=rtpmap:19 CN/8000
a=ptime:10
--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional
IAM,
GCI,acd52c00a3d511d9aa8a9d8cba1a49fa
--uniqueBoundary--
-*-
- 21 headers, 21 lines
* Using latest SIP request as basis request
* Sending to 66.178.100.66 : 5060 (NAT)
Apr 4 00:16:40 NOTICE[25296]: chan_sip2.c:5872
check_user_full: User name from URI: 66.178.100.66,
Digest auth user: (null)
== Authentication turned off, no secret for user
66.178.100.66
* No RDNIS header in SIP packet
-- - SIPFromURI:
;tag=8CB7504-1904
--> Transmitting (no NAT) response to
66.178.100.66:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Call-ID:
[EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
-*-
-- Executing Answer("SIP/66.178.100.66-bf34", "")
in new stack
* SDP preparation: We're at 62.56.250.198 port 17962
* Answering with preferred capability 0x2 (gsm)
* Answering with preferred capability 0x4 (ulaw)
* Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1(g723)
--> Reliably Transmitting (no NAT) response to
66.178.100.66:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Call-ID:
[EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 25296 25296 IN IP4 62.56.250.198
s=session
c=IN IP4 62.56.250.198
t=0 0
m=audio 17962 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
-*-
-- Executing Wait("SIP/66.178.100.66-bf34", "2")
in new stack
--- Sip read from 66.178.100.66:50341
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Date: Mon, 04 Apr 2005 00:16:50 GMT
Call-ID:
[EMAIL PROTECTED]
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK
-*-
- 9 headers, 0 lines
--- Sip read from 66.178.100.66:53065
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Date: Mon, 04 Apr 2005 00:16:50 GMT
Call-ID:
[EMAIL PROTECTED]
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 6
Timestamp: 1112573810
CSeq: 102 BYE
Content-Length: 0
-*-
- 11 headers, 0 lines
* Sending to 66.178.100.66 : 5060 (non-NAT)
--> Transmitting (no NAT) response to
66.178.100.66:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
-*-
== Spawn extension (AS5300, 9001, 2) exited non-zero
on 'SIP/66.178.100.66-bf34'
-- Executing Hangup("SIP/66.178.100.66-bf34", "")
in new stack
== Spawn extension (AS5300, h, 1) exited non-zero on
'SIP/66.178.100.66-bf34'
Destroying SIP dialogue '[EMAIL PROTECTED]'
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[Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP
hi's
i have been trying to configure my AS5300 to work with
my asterisk box. i have tried SIP, calls come,
answered and AS5300 sends BYE message after not more
than 5 secs. I have also tried MGCP, but i believe i
am not configuring that right. here is the output of
the sip debug. please help me out or lead me to the
direction of sorting this problem out.
thank you
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To:
Date: Mon, 04 Apr 2005 00:16:50 GMT
Call-ID:
[EMAIL PROTECTED]
Supported: timer
Min-SE: 600
Cisco-Guid:
2899651584-2748649945-2861211020-3122285050
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID:
;party=calling;screen=no;privacy=off
Timestamp: 1112573810
Contact:
Expires: 180
Allow-Events: telephone-event
MIME-Version: 1.0
Content-Type: multipart/mixed;boundary=uniqueBoundary
Content-Length: 431
--uniqueBoundary
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 5042 571 IN IP4
66.178.100.66
s=SIP Call
c=IN IP4 66.178.100.66
t=0 0
m=audio 18992 RTP/AVP 3 19
c=IN IP4 66.178.100.66
a=rtpmap:3 GSM/8000
a=rtpmap:19 CN/8000
a=ptime:10
--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional
IAM,
GCI,acd52c00a3d511d9aa8a9d8cba1a49fa
--uniqueBoundary--
-*-
- 21 headers, 21 lines
* Using latest SIP request as basis request
* Sending to 66.178.100.66 : 5060 (NAT)
Apr 4 00:16:40 NOTICE[25296]: chan_sip2.c:5872
check_user_full: User name from URI: 66.178.100.66,
Digest auth user: (null)
== Authentication turned off, no secret for user
66.178.100.66
* No RDNIS header in SIP packet
-- - SIPFromURI:
;tag=8CB7504-1904
--> Transmitting (no NAT) response to
66.178.100.66:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Call-ID:
[EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
-*-
-- Executing Answer("SIP/66.178.100.66-bf34", "")
in new stack
* SDP preparation: We're at 62.56.250.198 port 17962
* Answering with preferred capability 0x2 (gsm)
* Answering with preferred capability 0x4 (ulaw)
* Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1(g723)
--> Reliably Transmitting (no NAT) response to
66.178.100.66:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Call-ID:
[EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 25296 25296 IN IP4 62.56.250.198
s=session
c=IN IP4 62.56.250.198
t=0 0
m=audio 17962 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
-*-
-- Executing Wait("SIP/66.178.100.66-bf34", "2")
in new stack
--- Sip read from 66.178.100.66:50341
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Date: Mon, 04 Apr 2005 00:16:50 GMT
Call-ID:
[EMAIL PROTECTED]
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK
-*-
- 9 headers, 0 lines
--- Sip read from 66.178.100.66:53065
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Date: Mon, 04 Apr 2005 00:16:50 GMT
Call-ID:
[EMAIL PROTECTED]
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 6
Timestamp: 1112573810
CSeq: 102 BYE
Content-Length: 0
-*-
- 11 headers, 0 lines
* Sending to 66.178.100.66 : 5060 (non-NAT)
--> Transmitting (no NAT) response to
66.178.100.66:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.178.100.66:5060
From: ;tag=8CB7504-1904
To: ;tag=as08ade073
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
-*-
== Spawn extension (AS5300, 9001, 2) exited non-zero
on 'SIP/66.178.100.66-bf34'
-- Executing Hangup("SIP/66.178.100.66-bf34", "")
in new stack
== Spawn extension (AS5300, h, 1) exited non-zero on
'SIP/66.178.100.66-bf34'
Destroying SIP dialogue '[EMAIL PROTECTED]'
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Re: [Asterisk-Users] Zaptel Anti-MMX Optimizations
I put the Who? in Mishehu wrote: Did you try issuing "show translation recalc #" where # is any given number of seconds to recalculate for? For example, speex tends to show weird numbers for me on my dual proc xeon 2.8ghz, until I do a show translation recalc 1, then I get more sane numbers. I actually get really weird sounding calls when MXX_OPTIMIZATIONS are enabled as well as random crashes... so I don't think I need to worry how that table is calculated. It's just the best way I've found to qualify the problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with call-forward from ccme to * on sip trunk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks I've a strange problem, probably a mistake but I don't see it :( Description: My ephone-dn number on ccme, that is a simple connection plar for all ISDN calls, is 601 The voicemailmain on asterisk is 5900. CCME: 192.168.17.1 *: 192.168.17.10 My sip.conf: http://www.pastebin.com/266718 My extension.conf: http://www.pastebin.com/266720 My voicemail.conf: http://www.pastebin.com/266722 when I call the asterisk server from SIP free accounts, I receive the call on 601 (my 7960 phone) and then the call will be forwarded to voicemail without any problem. But when I receive a call from ISDN cloud, the 601 rings, the call is forwarded (see debug) on voicemail (number 5601), but the line goes down. This is the debug, that is I suppose the problem is on my Asterisk config (the 'ext-number' is the caller ID): http://www.pastebin.com/266724 I hope you could help me :) Thanks for all Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCUIYAMakHrsrHP9wRAlR1AKDKNzARotrmFMPphvjwqjp8da4SwACfQ6lo hxesZUu9t220j8zfQHW2DX0= =zJCw -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Authenticating username
> Dial(SIP/904)calls whoever logged on as john. You could define a variable in extensions.conf. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VG248 and Asterisk
Has anyone been successful getting a Cisco VG248 gateway to speak MGCP with Asterisk? If so can you share either your mgcp.conf or at least tips on getting the two devices working together. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to reset IAXy?
I'd put the device and another machine on a separate physical network where you can make whatever IP configurations you need in order to be able to send data to the IAXy. Then you can load new configuration to it there. There might be a better way to do i, but I don't know for sure. -mishehu Lam H. Nguyen wrote: Can anyone tell me how to reset the IAXy? I used I put it the wrong ip config in the IAXy and it conflicts with my network whenever I plug it in. Currently the DHCP is disable. I need to re-enable it to change the settings. The hard reset button on the IAXy doesn't seem to work __ Do you Yahoo!? Yahoo! Personals - Better first dates. More second dates. http://personals.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:424f6cce180855966097315! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SET & CHECK group
Look at:
http://www.voip-info.org/wiki-Asterisk+cmd+setgroup
read example 2 revised.
On Apr 3, 2005 1:20 PM, Mark Halverson <[EMAIL PROTECTED]> wrote:
> I attempted to use the incominglimit and outgoinglimit in iax.conf and it
> doesn't seem to work anylonger, running CVS-HEAD 3/16/05
>
> So I tried using the SetGroup but, in the dialplan I am already using Get
> and Check Group. I tried it with different variables and it still doesn't
> workany ideas?
>
> Basically I want each IAX Client coming in to be limited to a single call -
> I then only want 1 call on each outbound SIP channel/account.
>
> I believe the problem to be with (${CALLERIDNUM}) as it is setting the group
> to: CALLERID/something - that something is always changing with each call
> so the group using calleridnum never exceeds 1 and all calls go through.
>
> Example:
>
> exten => _1NXXNXX,1,SetGroup(${CALLERIDNUM})
> exten => _1NXXNXX,2,Checkgroup(1)
> exten => _1NXXNXX,3,SetGroup(CH1)
> exten => _1NXXNXX,4,CheckGroup(1)
> exten => _1NXXNXX,5,Dial(SIP/[EMAIL PROTECTED])
> exten => _1NXXNXX,103,background(busy)
> exten => _1NXXNXX,104,hangup
> exten => _1NXXNXX,105,SetGroup(CH2)
> exten => _1NXXNXX,106,CheckGroup(1)
> exten => _1NXXNXX,107,Dial(SIP/[EMAIL PROTECTED])
> exten => _1NXXNXX,207,Dial(IAX2/[EMAIL PROTECTED])
>
> --
> No virus found in this outgoing message.
> Checked by AVG Anti-Virus.
> Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 4/1/2005
>
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[Asterisk-Users] IAX messages
I'm trying to get IAXTEL inbound working in my log I'm seeing all this noise (below). I understand I'm in DEBUG mode but I'm not doing anything yet ... what do all these messages mean??? Apr 2 02:47:44 DEBUG[28339]: Immediately destroying 2, having received INVAL Apr 2 02:47:44 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=941 Apr 2 02:48:44 DEBUG[28339]: Immediately destroying 1, having received INVAL Apr 2 02:49:32 DEBUG[28339]: Sending VNAK Apr 2 02:49:32 DEBUG[28339]: Sending VNAK Apr 2 02:49:32 DEBUG[28339]: Sending VNAK Apr 2 02:49:34 DEBUG[28339]: Sending VNAK Apr 2 02:49:34 VERBOSE[28339]: -- Registered to '69.73.19.178', who sees us as 68.52.23.171:19548 Apr 2 02:49:34 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=1330 Apr 2 02:49:34 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=1330 Apr 2 02:49:34 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=1330 Apr 2 02:49:34 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=1330 Apr 2 02:50:23 DEBUG[28339]: Immediately destroying 1, having received INVAL Apr 2 02:50:23 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047 Apr 2 02:50:26 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047 Apr 2 02:50:33 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047 Apr 2 02:50:35 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047 Apr 2 02:50:36 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=1047 Apr 2 02:51:22 DEBUG[28339]: Immediately destroying 2, having received INVAL Apr 2 02:52:04 DEBUG[28339]: Immediately destroying 1, having received INVAL Apr 2 02:52:04 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=1, dst=68 Apr 2 02:52:38 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=336 Apr 2 02:52:49 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=336 Apr 2 02:52:59 DEBUG[28339]: Raw Hangup 69.73.19.178:4569, src=2, dst=336 Apr 2 02:53:44 DEBUG[28339]: Immediately destroying 1, having received INVAL ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls
On Apr 3, 2005 5:45 PM, John Novack <[EMAIL PROTECTED]> wrote: > Well, you COULD use your delete key. Actually nope, I can't because I'm using gmails web client to read my email. > You DO have one, don't you? Yep I do, how did you know? > And you complain of others posting stupidity Please read what I answered about NOT being able to use the deleted key, then read the line about complaing.. then look in the mirror. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel Anti-MMX Optimizations
Did you try issuing "show translation recalc #" where # is any given number of seconds to recalculate for? For example, speex tends to show weird numbers for me on my dual proc xeon 2.8ghz, until I do a show translation recalc 1, then I get more sane numbers. Just my thoughts. -mishehu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI/Dialing out
Hi, Philip Hofstetter wrote: Now may next step has been to enable dialing out with the softphones. This does not work as expected. I was able to fix this problems by downgrading from kernel 2.6.11 to 2.6.10. There must be a CAPI-Problem hidden somewhere. Last saturday was so much fun for me, trying out all the stuff that can be done with asterisk. Thanks to all for this wonderful program! Philip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
iptables looks very powerful, thats for sure. I prefer PF's approach to security first, convenience second, and I *really* like the fact that PF has a real parser. As the requements get more complex, having everything in one file, and very readable and structured is a huge plus. Also, the integration with ALTQ is nice, especially for these types of applications. Andrew Kohlsmith wrote: On April 3, 2005 08:13 am, Tim Pushor wrote: To someone who has never installed OpenBSD (or FreeBSD + pf for that matter) the learning curve is going to be much much higher than 15 minutes, although one you learn PF you will never go back! I've never seen the great advantage to pf over ip and tc. Perhaps I'm just not that learned though. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, April 03, 2005 3:33 AM Subject: [Asterisk-Users] Router with QoS recommendations As I have a Cisco PIX 515, with NO QoS functionality, and I'm looking for a router that does outgoing QoS to put in front of my PIX. PixOS 7.0.1 supports QoS. Yesterday it was on TAC's download page. No, I have not installed yet. I.N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] really small box
Hello, Matt! MR> fine. If you have to do any sort of transcoding a soekris is not the MR> way to go but for a small installation it works great. Well.. Cisco's 17xx series router is a device which you can take, plug, configure and have office PBX. But price tag is $2K. Why the same can't be done for a fraction of this price using * and not involving active cooling and graphics cards? 20-30 office users + 3-4 transcoding sessions + voicemail. What kind of horsepower do you need for this? MR> I run an entire asterisk installation off of a 512 MB CF card (have MR> ~250 MB to spare for voicemails and logs) Do you have install/configuration/HOWTO document? If yes, could you post it here or just send it to mail email? I.N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls
Well, you COULD use your delete key. You DO have one, don't you? And you complain of others posting stupidity JN C F wrote: What can be done to this shmuck? Everytime I post anything to the list I get one of these. I'm sure I'll get one for posting this one as well. -- Forwarded message -- From: [EMAIL PROTECTED] <[EMAIL PROTECTED]> Date: Apr 3, 2005 12:24 AM Subject: NDN: Re: [Asterisk-Users] Delaying answer of incoming calls To: C F <[EMAIL PROTECTED]> Sorry. Your message could not be delivered to: Joshua Chessman (Mailbox or Conference is full.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Discussion Form
One would hope so, but one of the fist posts I see is someone ranting on about how if you haven't read x or y you don't deserve an answer. It is that kind of a social misfit that should not be welcome anywhere, but seems to have too loud a voice here. JN Ty Carter wrote: Thank you for your contribution Now maybe this is a good place where people can ask a question without getting slammed because they don't understand their own ignorance. :-) [EMAIL PROTECTED] wrote: List: With recent discussions in regards to a forum, I have set-up a multi-faceted Asterisk and Open Source Discussion Board. The link is www.voipnewbie.com/forum It is open and ready for use. Enjoy! VoIPNewbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packetization
I have to admit this still doesn't make sence.. if sipura's default is .03ms and asterisk is 20ms.. why is the sipura dumping out around 60 frames/sec while the sipura is dumping out around 30 frames/sec?? Shouldn't the frames / packets per second go UP as the packetization gets smaller?On Apr 3, 2005 5:25 PM, Matt <[EMAIL PROTECTED]> wrote:Never mind... blah spoke before I thought :P Found the setting On Apr 3, 2005 5:23 PM, Matt <[EMAIL PROTECTED]> wrote:Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue (audio going from the phone over wireless is slightly choppy).. while audio coming in (20ms) is ok... where do you change it on the sipura?On Apr 3, 2005 4:07 PM, Bruce Komito <[EMAIL PROTECTED]> wrote:The packet size is a function of the number of milliseconds of sound sentin the RTP packet. I don't know how to force * to change this, but you*can* unilaterally change the RTP packet size on the Sipura. By doingthis, RTP packets sent by the Sipura will be larger or smaller than thedefault (.03 ms is the default), and I know * will swallow whatever theSipura sends it. So, I know it's possible to change this in at least onedirection if you are using a Sipura.Bruce KomitoHigh Sierra Networks, Inc.www.servers-r-us.com(775) 236-5815On Sun, 3 Apr 2005, Matt wrote:> IAX is not an option as Sipura devices do not support AIX.> Yes, the sipura will handle the different packet sizes...>> Is it possible to reprogram asteris to do this?>> On Apr 3, 2005 1:55 AM, Steven Critchfield <[EMAIL PROTECTED]> wrote:> >> > On Sat, 2005-04-02 at 21:16 -0500, Matt wrote:> > > I'm aware that asterisk only supports 20ms packetization rates. Due> > > to the fact that I will be using some voip devices on a wireless> > > network which is highly sensative to framerate.. is there any way I> > > can hard code the packetization rate at say 30 or 40ms and then> > > compile astrisk? If so, can anyone in the know tell me what variables> > > I need to look at to change?> >> > Are you sure your other devices support different packet sizes? Are you> > sure the added delay in audio delivery can be handled decently and not> > cause added echo?> >> > Have you considered what IAX trunking can do for you? It will reduce> > frame rate as you add channels since each packet will then hold the> > frames for each of the consecutive calls.> > --> > Steven Critchfield <[EMAIL PROTECTED]>> >> This message has been categorized as "Indeterminate" by Bayesian Analyzer.> Please click on this link if this message is a Spam> http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=2>> Or on this link if this message is a legitimate mail> http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=1>>> --> ---> This message has been inspected by DynaComm i:mail> ---> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packetization
Never mind... blah spoke before I thought :P Found the setting On Apr 3, 2005 5:23 PM, Matt <[EMAIL PROTECTED]> wrote:Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue (audio going from the phone over wireless is slightly choppy).. while audio coming in (20ms) is ok... where do you change it on the sipura?On Apr 3, 2005 4:07 PM, Bruce Komito <[EMAIL PROTECTED]> wrote:The packet size is a function of the number of milliseconds of sound sentin the RTP packet. I don't know how to force * to change this, but you*can* unilaterally change the RTP packet size on the Sipura. By doingthis, RTP packets sent by the Sipura will be larger or smaller than thedefault (.03 ms is the default), and I know * will swallow whatever theSipura sends it. So, I know it's possible to change this in at least onedirection if you are using a Sipura.Bruce KomitoHigh Sierra Networks, Inc.www.servers-r-us.com(775) 236-5815On Sun, 3 Apr 2005, Matt wrote:> IAX is not an option as Sipura devices do not support AIX.> Yes, the sipura will handle the different packet sizes...>> Is it possible to reprogram asteris to do this?>> On Apr 3, 2005 1:55 AM, Steven Critchfield <[EMAIL PROTECTED]> wrote:> >> > On Sat, 2005-04-02 at 21:16 -0500, Matt wrote:> > > I'm aware that asterisk only supports 20ms packetization rates. Due> > > to the fact that I will be using some voip devices on a wireless> > > network which is highly sensative to framerate.. is there any way I> > > can hard code the packetization rate at say 30 or 40ms and then> > > compile astrisk? If so, can anyone in the know tell me what variables> > > I need to look at to change?> >> > Are you sure your other devices support different packet sizes? Are you> > sure the added delay in audio delivery can be handled decently and not> > cause added echo?> >> > Have you considered what IAX trunking can do for you? It will reduce> > frame rate as you add channels since each packet will then hold the> > frames for each of the consecutive calls.> > --> > Steven Critchfield <[EMAIL PROTECTED]>> >> This message has been categorized as "Indeterminate" by Bayesian Analyzer.> Please click on this link if this message is a Spam> http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=2>> Or on this link if this message is a legitimate mail> http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=1>>> --> ---> This message has been inspected by DynaComm i:mail> ---> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email forum??
On Mar 31, 2005 11:26 AM, Chuck Bunn <[EMAIL PROTECTED]> wrote: > I am new to Asterisk and the first thing I have noticed about Asterisk > and Pingtels open PBX's is that they are using this dinosaur method of > running forums. It is a real pain getting every message in the forum and > essentially keeping my own database of issues. With that said are there > any forums that are well used or that might even convert this email in a > true forum that is searchable and that doesn't require me downloading > every email. Before you go and rant on me go see how Mambo Server does > it at http://forum.mamboserver.com. The forums are easy to use and thus > are easy to participate in. I use mozilla Thunderbird and I have setup > filters and all but it still is a pain to use this outdated email forum. If you are looking for a web interface to this mailing list, get yourself a Gmail account and subscribe from it. It does fantastic threading. If you need an account, I'll be happy to help you out, I have 50 invites. -- Leif Madsen http://www.leifmadsen.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packetization
Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue (audio going from the phone over wireless is slightly choppy).. while audio coming in (20ms) is ok... where do you change it on the sipura?On Apr 3, 2005 4:07 PM, Bruce Komito <[EMAIL PROTECTED]> wrote:The packet size is a function of the number of milliseconds of sound sentin the RTP packet. I don't know how to force * to change this, but you*can* unilaterally change the RTP packet size on the Sipura. By doingthis, RTP packets sent by the Sipura will be larger or smaller than thedefault (.03 ms is the default), and I know * will swallow whatever theSipura sends it. So, I know it's possible to change this in at least onedirection if you are using a Sipura.Bruce KomitoHigh Sierra Networks, Inc.www.servers-r-us.com(775) 236-5815On Sun, 3 Apr 2005, Matt wrote:> IAX is not an option as Sipura devices do not support AIX.> Yes, the sipura will handle the different packet sizes...>> Is it possible to reprogram asteris to do this?>> On Apr 3, 2005 1:55 AM, Steven Critchfield <[EMAIL PROTECTED]> wrote:> >> > On Sat, 2005-04-02 at 21:16 -0500, Matt wrote:> > > I'm aware that asterisk only supports 20ms packetization rates. Due> > > to the fact that I will be using some voip devices on a wireless> > > network which is highly sensative to framerate.. is there any way I> > > can hard code the packetization rate at say 30 or 40ms and then> > > compile astrisk? If so, can anyone in the know tell me what variables> > > I need to look at to change?> >> > Are you sure your other devices support different packet sizes? Are you> > sure the added delay in audio delivery can be handled decently and not> > cause added echo?> >> > Have you considered what IAX trunking can do for you? It will reduce> > frame rate as you add channels since each packet will then hold the> > frames for each of the consecutive calls.> > --> > Steven Critchfield <[EMAIL PROTECTED]>> >> This message has been categorized as "Indeterminate" by Bayesian Analyzer.> Please click on this link if this message is a Spam> http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=2>> Or on this link if this message is a legitimate mail> http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=1>>> --> ---> This message has been inspected by DynaComm i:mail> --->___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Suse minimal installation based on Suse Rescue - what to add to be bootable on HD partition ?
Hi, I'm trying to go route some of Asterisk users already proposed for Asterisk minimal system. I've started from Suse Rescue system image - I've put it into HD partition. But since rescue is spawned from working system it has empty /boot directories and is not directly bootable if put on HD. I've tried to transfer or install kernel and grub to this partition, but no success (I first access to partition with chroot to make additions...). I get errors on kernel rpm -ivh installation (I guess there are no directories and dependencies found in chroot) and also get error on "grub-install /dev/hda7": Could not find device for /boot: Not found or not a block device Is there anyone more experienced with some advice, howto or example what need to be added to partition to be bootable on HD ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Buying some Polycom IP300s
We have a majority of IP300's, and a few IP500's. The IP300's are great phones if you need to simply drop in a bunch of VoIP phones quickly and cheaply. The IP300's simply lack certain features like speakerphone that you may want. Aside from that, its a great phone. -Courtney Dan Morin wrote: Sorry for the double post, I tried to paste and accidently sent the email I've been playing with Asterisk for a few weeks now, and I've gotten everything to work well with softphones, so I'm ready to move on to normal VoIP phones. I've been looking around and reading comments that people have had, and I was convinced that the Polycom IP300 was a great phone for a good price. But, then I ran into this page, which has been update in the last few days: http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500 The page in the wiki used to say that the person would not recomed Polycom phones to anyone. So anyway, I just want to make sure that the IP300 is a good choice. I don't want to get cheap phones that aren't business quality, since I do play on using them for my business after testing. Also, is the IP500 worth the extra money? What can it do that the IP300 can't. And finally, will the IP300 do ulaw encoding? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting when a called mobile is not reachable?
This is traditional accross the mobile / cell providers, and there is no real way around it. Background : The only way to ensure that a mobile is truly there is to page the mobile, normally based on the Mobile Switching Centre (MSC) coverage area, and thats after looking up on the subscirbers HLR, its a lot of signalling for a call not to connect, and a cost to the operator. With the rate that mobile operators charge the A party for the call, they get a percentage of the call from the originating operator, so they get cash as soon as it connects, and therefor its in their interest to connect that call, even if its to an announcement shelf. Its one of the reasons they invented voicemail If there is a way around it, don't shout it too loudly David On Apr 3, 2005 8:56 PM, Ian Hailey <[EMAIL PROTECTED]> wrote: > Hello all, > > I was hoping to be able to call a mobile and if it is un-reachable for > whatever reason (e.g. switched off) then I was expecting an unobtainable > response that would be detected in Asterisk. It seems that the operator > (Virgin in UK) imedately completes the call and plays an automated > message before clearing the call. Does anyone know if there a way of > avoiding the call completion for mobiles? I have noticed that Sipgate > charge for a calls to an unavailable mobile regardless. > > Thanks. > > Ian. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for res_config_pgsql
A google search shows exactly one reference, so it appears to exist somewhere. It's in somebodies CVS, any ideas? -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packetization
The packet size is a function of the number of milliseconds of sound sent in the RTP packet. I don't know how to force * to change this, but you *can* unilaterally change the RTP packet size on the Sipura. By doing this, RTP packets sent by the Sipura will be larger or smaller than the default (.03 ms is the default), and I know * will swallow whatever the Sipura sends it. So, I know it's possible to change this in at least one direction if you are using a Sipura. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sun, 3 Apr 2005, Matt wrote: > IAX is not an option as Sipura devices do not support AIX. > Yes, the sipura will handle the different packet sizes... > > Is it possible to reprogram asteris to do this? > > On Apr 3, 2005 1:55 AM, Steven Critchfield <[EMAIL PROTECTED]> wrote: > > > > On Sat, 2005-04-02 at 21:16 -0500, Matt wrote: > > > I'm aware that asterisk only supports 20ms packetization rates. Due > > > to the fact that I will be using some voip devices on a wireless > > > network which is highly sensative to framerate.. is there any way I > > > can hard code the packetization rate at say 30 or 40ms and then > > > compile astrisk? If so, can anyone in the know tell me what variables > > > I need to look at to change? > > > > Are you sure your other devices support different packet sizes? Are you > > sure the added delay in audio delivery can be handled decently and not > > cause added echo? > > > > Have you considered what IAX trunking can do for you? It will reduce > > frame rate as you add channels since each packet will then hold the > > frames for each of the consecutive calls. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > > > > This message has been categorized as "Indeterminate" by Bayesian Analyzer. > Please click on this link if this message is a Spam > http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=2 > > Or on this link if this message is a legitimate mail > http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=1 > > > -- > --- > This message has been inspected by DynaComm i:mail > --- > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting when a called mobile is not reachable?
Hello all, I was hoping to be able to call a mobile and if it is un-reachable for whatever reason (e.g. switched off) then I was expecting an unobtainable response that would be detected in Asterisk. It seems that the operator (Virgin in UK) imedately completes the call and plays an automated message before clearing the call. Does anyone know if there a way of avoiding the call completion for mobiles? I have noticed that Sipgate charge for a calls to an unavailable mobile regardless. Thanks. Ian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Buying some Polycom IP300s
Dan Morin wrote: > Sorry for the double post, I tried to paste and accidently sent the > email > > I've been playing with Asterisk for a few weeks now, and I've gotten > everything to work well with softphones, so I'm ready to move on to > normal VoIP phones. I've been looking around and reading comments > that people have had, and I was convinced that the Polycom IP300 was > a great phone for a good price. But, then I ran into this page, > which has been update in the last few days: > > http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500 > > The page in the wiki used to say that the person would not recomed > Polycom phones to anyone. So anyway, I just want to make sure that > the IP300 is a good choice. I don't want to get cheap phones that > aren't business quality, since I do play on using them for my > business after testing. Also, is the IP500 worth the extra money? > What can it do that the IP300 can't. And finally, will the IP300 do > ulaw encoding? The IP300 is a nice entry-level business phone. It does not have a speakerphone, and cannot handle PoE, but other than that it is excellent. It is more expensive than some of the fully-featured generic phones, but it also is built to a much higher standard, including a properly weighted handset and high impact plastic. If price is the main thing, then this phone might be a bit too expensive ($130-$150), but if quality (or even just the *feeling* of quality) is important, this phone will serve well. The IP500 is a similar phone with more line appearances, a higher resolution display, full handsfree (Polycom-quality) and PoE. The IP500 has been favorably compared to the Cisco 7940. Cheers, -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 01/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA 2000 - Miltiple Ring Tones
Rod Bacon wrote: I'm glad I'm not the only one Now... for a solution? Well at least this rules out a misconfiguration on the telco's end (unless both our telco's made the same mistake). Does /anyone/ at all have any suggestions, or is there some debug information we can send to the list to figure this out? It is quite annoying :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom and Multiple calls
Hi! > On the snom (I've tested this on the 220 and 360), the phone will > immediately reject any new INVITE that arrives with 486 BUSY HERE if > there's already a call on the phone opening That is very interesting - can you present a review of the Snom 360 hardware, even if it is a short one? Possibly compare it to the 220 and illustrate the differences? Some questions: - quality of the handset and the speaker phone? - how do the buttons feel? - are the line LEDs multi-colour ones? As the 360 softphone is available to everyone the software part isn't _that_ interesting as I can play with it myself... :-) Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Authenticating username
Hi, >From what I can see in the documentation the title of the section in sip.conf is the username that the user logs in as. Is there a way of seperating the names so that you can login with a normal username, but call them with SIP/extension. Like so: [904] authuser=john secret=password etc... Dial(SIP/904)calls whoever logged on as john. Any ideas? -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Passing varibles *out* of macros
Have you tried putting in some NoOp lines to verify the values of
${screenresult}?
Also, wouldn't you get the desired result by removing the 'g' option
from your Dial()?
You might want to add an 'h' extension for further processing on the
dead channel.
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Re: [Asterisk-Users] Router with QoS recommendations
On April 3, 2005 08:13 am, Tim Pushor wrote: > To someone who has never installed OpenBSD (or FreeBSD + pf for that > matter) the learning curve is going to be much much higher than 15 > minutes, although one you learn PF you will never go back! I've never seen the great advantage to pf over ip and tc. Perhaps I'm just not that learned though. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
NVC List Manager wrote: As usual there's nothing that will beat OpenBSD. Takes 15 minutes to build following the instructions on the CD cover. To someone who has never installed OpenBSD (or FreeBSD + pf for that matter) the learning curve is going to be much much higher than 15 minutes, although one you learn PF you will never go back! Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
On Sunday 03 April 2005 06:33, [EMAIL PROTECTED] wrote: > Hi List > > > > As I have a Cisco PIX 515, with NO QoS functionality, and I'm looking for a > router that does outgoing QoS to put in front of my PIX. Problem is that > I'm using my 768/8096Kbit ADSL for both data and VoIP, and as soon as data > is being sent to the internet the sound quality drops to something that is > of NO use. > > > > Any suggestions or recommendations is appreciated. As usual there's nothing that will beat OpenBSD. Takes 15 minutes to build following the instructions on the CD cover. -- NVC List Manager (For external lists) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Passing varibles *out* of macros
An option, but what about multiple inbound calls? I'd be worried that they trip over each other. But - given the odds of this happening (variable is set and then read instantly) - it may be the route to go. Thanks - Joe > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Wilson Pickett > Sent: Sunday, April 03, 2005 5:35 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Passing varibles *out* of macros > > How about setGlobalVar() > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
Hi! > As I have a Cisco PIX 515, with NO QoS functionality, and Im looking for > a router that does outgoing QoS to put in front of my PIX. Problem is > that Im using my 768/8096Kbit ADSL for both data and VoIP, and as soon > as data is being sent to the internet the sound quality drops to > something that is of NO use. > > Any suggestions or recommendations is appreciated. Checkout m0n0wall on a Soekris or WRAP device. Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packetization
IAX is not an option as Sipura devices do not support AIX.Yes, the sipura will handle the different packet sizes... Is it possible to reprogram asteris to do this? On Apr 3, 2005 1:55 AM, Steven Critchfield <[EMAIL PROTECTED]> wrote:On Sat, 2005-04-02 at 21:16 -0500, Matt wrote:> I'm aware that asterisk only supports 20ms packetization rates. Due> to the fact that I will be using some voip devices on a wireless> network which is highly sensative to framerate.. is there any way I> can hard code the packetization rate at say 30 or 40ms and then> compile astrisk? If so, can anyone in the know tell me what variables> I need to look at to change?Are you sure your other devices support different packet sizes? Are yousure the added delay in audio delivery can be handled decently and notcause added echo?Have you considered what IAX trunking can do for you? It will reduceframe rate as you add channels since each packet will then hold theframes for each of the consecutive calls.--Steven Critchfield <[EMAIL PROTECTED]>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SET & CHECK group
I attempted to use the incominglimit and outgoinglimit in iax.conf and it
doesn’t seem to work anylonger, running CVS-HEAD 3/16/05
So I tried using the SetGroup but, in the dialplan I am already using Get
and Check Group. I tried it with different variables and it still doesn't
workany ideas?
Basically I want each IAX Client coming in to be limited to a single call -
I then only want 1 call on each outbound SIP channel/account.
I believe the problem to be with (${CALLERIDNUM}) as it is setting the group
to: CALLERID/something - that something is always changing with each call
so the group using calleridnum never exceeds 1 and all calls go through.
Example:
exten => _1NXXNXX,1,SetGroup(${CALLERIDNUM})
exten => _1NXXNXX,2,Checkgroup(1)
exten => _1NXXNXX,3,SetGroup(CH1)
exten => _1NXXNXX,4,CheckGroup(1)
exten => _1NXXNXX,5,Dial(SIP/[EMAIL PROTECTED])
exten => _1NXXNXX,103,background(busy)
exten => _1NXXNXX,104,hangup
exten => _1NXXNXX,105,SetGroup(CH2)
exten => _1NXXNXX,106,CheckGroup(1)
exten => _1NXXNXX,107,Dial(SIP/[EMAIL PROTECTED])
exten => _1NXXNXX,207,Dial(IAX2/[EMAIL PROTECTED])
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 4/1/2005
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Re: [Asterisk-Users] SIP dialing in two extensions
Jozeph Brasil a écrit : Hi guys, Is it possible to make Dial to call two extensions at the same time? I want when the user pressed extension it call to two SIP phones at the same time... Who wakeup first get the call... Dial(SIP/&IAX2/&OH323/<...>) -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Welcome to the Asterisk users community! Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Asterisk.org is a fast moving project. New code is added every day. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. Again, welcome to the Asterisk.org Open Source PBX Project! Astricon Europe registration is new open! http://www.astricon.net Meet you on the IRC channel :-), the bug tracker or on the mailing list! /oej ** Asterisk version information At this moment we have two current versions of Asterisk, the developer version and the stable version. The stable version is distributed as .tar.gz archives on several servers. The current stable version of Asterisk is 1.0.7. The stable version contains no new functions and only changes when bugs are fixed. The development version is to be used by people that can test new functions and live with bugs and unexpected shortcomings. The development version is branded 1.1 and will be the basis for the next stable version, version 1.2. We will hopefully soon reach a code freeze and start testing the stability of version 1.1, so we will need your help. ** The mailing list is growing Today, we propably have over 10,000 readers on the -users list. This means that everything anyone write to this mailing list, is sent to thousands of mailboxes that are already flowing over with messages. That's why we all need to follow some simple rules on how to use the mailing list and the other tools that are available. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. And please do not send out "test" messages to the list. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook "The hitchhiker's guide to Asterisk" is already well worth reading. * Asterisk Daily news is at http://www.sineapps.com/news.php * VoIP-search (Asterisk mailing list etc) http://search.voip-forum.com Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. Do not use this list as a secondary support line if you do not get an answer on the -users list. It is meant for developer discussions, not advanced support. If you need answers, there is a better chance that you will get help on the irc channel. For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a list called asterisk-bsd. There is also a business list for those that want to ask for commercial services and inform their community about new services (asterisk-biz). You'll find all lists on http://lists.digium.com, which is the site where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. If you are unsure which list to use, send only to the -users list. Make sure that you remove unnecessary text when you reply, to make it easy to browse the mailing list quickly. And please do not send HTML mail to a mailing list. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new bug; often, your bug or feature already exists but is simply slowly making it's way through the system. Duplicate re
[Asterisk-Users] SIP dialing in two extensions
Hi guys, Is it possible to make Dial to call two extensions at the same time? I want when the user pressed extension it call to two SIP phones at the same time... Who wakeup first get the call... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email forum??
Hi I haven't read all of the messages in this lengthy thread, so I hope I'm not repeating something from it. Just a couple of questions: 1. What about mail-archive.com for archiving the list? 2. The archive need not be related to the list. It just needs to be "subsribed to it. Anybody want to set up searchable archives for the list? 3. What about web-forum <-> mailing list gateways? E.g: http://www.phorum.org (never used it, just heard about it). 4. I'll just mention again that this list has a very high load. Can anybody suggest separate topics that could be moved to "sublists"? -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Voice mail with CCM
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Apr 2, 2005, at 8:00 PM, Nathan Alberti wrote: I'm currently in the process of getting it to work for a CCME install, I have it all working except for one thing.. I think it was calling a phone from the asterisk server the call transfer back to asterisk would fail with an authentication issue and die. I'm pretty sure this issue can be resolved I just have not had the time recently wo work on it, I can provide more info when I'm back in the office next week. If you have any question, please ask :) I've a solution like that in production, with a SIP trunk between ccme and *, without problems. The MWI works too. Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCUBuTMakHrsrHP9wRAvk9AKC1YqetsRZXw1wrOKXrqemSwFxDOACdFjS1 vpGDh7BzIUwDAQBnwMwzVq8= =C1ka -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Jasomi Peerpoing
Hi Im having Jasomi peerpoint far end SBC & im trying to integrate this with asterisk . When i call any no it directly goes to his voice mail. But when i start debug on asterisk it received 403 Forbidden Proxy OutBound Policy from Peerpoint and call is not working . is anybody using asterisk with Session Border Controller ? sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060)bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)realm=Asterisk ; Our global authentication realmsrvlookup = yesvideosupport = yesdisallow = allallow = gsmallow = ulawallow = alawallow = h261allow = h263 sip_additional.conf [7101]username=7101type=friendsecret=xxx qualify=1000port=5060nat=yesmailbox=7101host=dynamicdtmfmode=rfc2833context=from-sipcanreinvite=yescallerid=" Dhananjay S" <7101> i tried with both canreinvite=yes and no but fails. Thanks in advance Regards Dhananjay Sarnaik Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looping messages
Chris Blake wrote: Greetings *`s, I have set up a call which constantly loops a pre-recorded message waiting for the user to press a digit on their phone. At this point the call is sent elsewhere in the dialplan. But if the called party doesn`t press any buttons and hangs up, the message carries on playing...the same goes for if the called party hangs up without pressing any buttons. The same happens if the call goes thru to the called party`s voicemail..it plays the message but doesn`t stop. Here is the section in my dialplan : [realyst1] exten => s,1,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,2,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten => s,3,Answer exten => s,4,Wait(1) exten => s,5,Background(realyst/updaterequest) ; "play outbound msg" exten => s,6,Background(realyst/acknowledge) ; "Press 1 to replay or 2 to acknowledge receiving this message" exten => s,7,Goto(s,5) exten => 1,1,Goto(s,5) ; replay message exten => 2,1,Goto(msgack,s,1) ; acknowledge message exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup Any links/ideas/tips welcome... Regards -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] Remember that as a teenager you are in the last stage of your life when you will be happy to hear that the phone is for you. -- Fran Lebowitz, "Social Studies" ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In this case it won't timeout and will go into an endless loop, maybe if u use the h extension to detect hangup Ezabi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Macro Extension with Realtime and Mysql DB
Thank Matthew:
I do that, i create the database with tables for support RT Asterisk, then i
create the context deafult in the database, but the macro that i use is
steel in the etension.conf and its works.
Database Extension:
IDCONTEX EXTENPRIORITYAPP APPDATA
1 default _2XX 1 Macro
test1|SIP/${EXTEN:0}
2 default _3XX 1 Macro
test1|SIP/${EXTEN:0}
3 default _4XX 1 Macro
test1|SIP/${EXTEN:0}
Extension.conf:
[default]
switch => Realtime/default@
[macro-test1]
exten => s,1,Dial(${ARG1},20,tTr)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => s-BUSY,2,Goto(default,s,1)
exten => _s-.,1,Goto(s-NOANSWER,1)
exten => a,1,VoicemailMain([EMAIL PROTECTED])
Its works, but if i change the macro test to database, its doesn't works, my
database and extensions.conf loooks like:
Database Extension:
IDCONTEX EXTEN PRIORITYAPPAPPDATA
1 default _2XX 1 Macro
test1|SIP/${EXTEN:0}
2 default _3XX 1 Macro
test1|SIP/${EXTEN:0}
3 default _4XX 1 Macro
test1|SIP/${EXTEN:0}
4 test1 s 1 Dial
${ARG1}|20|tTr
5 test1 s 2 Goto
s-${DIALSTATUS}|1
6 test1 s-NOANSWER 1Voicemail
u${MACRO_EXTEN}
7 test1 s-NOANSWER 2Goto
default|s|1
8 test1 s-BUSY1Voicemail
b${MACRO_EXTEN}
9 test1 s-BUSY2Goto
default|s|1
10test1 _s-. 1Goto
s-NOANSWER|1
11test1 a1 VoicemailMain
[EMAIL PROTECTED]
And the extensions.conf looks:
[default]
switch => Realtime/default@
[macro-test1]
switch => Realtime/test1@
The error on CLI Asterisk is the context macro-test1 no exist for macro
test1
But, this configuration don't work.
Any idea.
Thank.
Kritikus.
ehm <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users
Subject: Re: [Asterisk-Users] Macro Extension with Realtime and Mysql DB
Date: Sat, 02 Apr 2005 23:04:01 -0600
AFAIK, you would configure a macro extension in RealTime just like you
configure a regular extension/context in RealTime.
-Matthew
> From: kritikus Araklidas <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: Sun, 03 Apr 2005 04:11:38 +
> To:
> Subject: [Asterisk-Users] Macro Extension with Realtime and Mysql DB
>
> Hi Everyone:
>
> I need to know if somebody know how to configure macro extension
> (extension.conf) in the database for Asterisk Realtime support if is
> suported.
>
> Regards,
>
> Kritikus
>
> _
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Re: [Asterisk-Users] Asterisk Auto-Startup on Ubuntu/Debian
On Sat, Apr 02, 2005 at 01:20:37PM -0500, Josh Alberts wrote: > I'm having trouble getting asterisk to run at startup using Ubuntu. > I've checked, and the asterisk dameon is set to run at init 5. However, > I'm not seeing anything that says that asterisk has been started during > the boot process. Oddly, when I shut the machine down/run init6, it > says Starting Asterisk PBX. Odd. I'm using the default scripts that > came with asterisk (I installed using synaptic and the debian universe > repositories). What version of Asterisk is that, BTW? AFAIK the ubuntu package is an older version of the current Debian package. > I've edited /etc/default/asterisk, uncommented the first > line and changed start asterisk to "yes". Anybody know what might be > wrong? ls -l /etc/rc?.d/*asterisk man update-rc.d But generally the package's init script is automatically being added, and will silently exit if you configured the service not to run or if your system does not have the binary. Anyway, maybe Asterisk has started but has failed to load? -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Registration to multiple GKs
Charles, I don't think asterisk is a full GK. So if you are asking if asterisk will send out LRQ to the neighbors then I don't believe it would. As far as registering with multiple gk, I wanted to correct myself. An endpoint/gw can register with one primary gk and a number of backup gk. If the primary gk fails, then request will be sent to backup gk in the order of registration. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang Sent: Sunday, April 03, 2005 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Registration to multiple GKs Is it possible to run Asterisk with another GKs using Neighbor mode? If it is possible, we can run asterisk with several gnugks. On Apr 2, 2005 10:41 PM, Alex Vishnev <[EMAIL PROTECTED]> wrote: > I don't think you can. The rules of h323 is so that you can register with a > single gk at a time. > > Alex > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie > Sent: Saturday, April 02, 2005 6:37 AM > To: Asterisk-Users@lists.digium.com > Subject: [Asterisk-Users] Registration to multiple GKs > > Hi all, > > How can I configure chan_h323 or chan_oh323 to register to multiple GK > and route calls in-between? > > Many thanks. > Newbie > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open Source Billing Software
I can say that I use FPDF.org for my OSRAIDS project. Take a look at how I create PDFs on the fly. http://OSRAIDS.org On Mar 31, 2005 1:15 PM, Max W Blackmer Jr <[EMAIL PROTECTED]> wrote: > I am just beginning work on Trabas now. nothing as of yet. I just liked > the features that it currently offers, but it does definitely need > allot of work yet. I am looking at adapting this one or take the > concepts and rewrite for PHP. > > Some features I am looking for that are not in the current system. > > 1. Better ability to pull in records from asterisks CDR using billing > codes. > 2. Dynamic reports for CDR according to Clients requirements. > 3. Allow clients to look at the current state of their account to > integrate to End user web site through SOAP calls. > 4. Make PDF bills and Reports with the capability of emailing or > generate on demand for web download for clients. > > Any other Ideas anyone might need in addition to trabas features? > > Thanks, > > Max W. Blackmer, Jr. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Andrew Latham http://www.lathama.com [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] If any of the above are not working, we have bigger problems than my email. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] xlite regestration fails but calls to thru
Scott, First, you need to get the most recent os for the pix, otherwise you will have a lot of problems with udp packets and translations (including bad checksum on your udp packets). I am running both pix515 and pix501 without a problem with sip and h323. you don’t need to open any ports on the pix, because the firewall is an ALG( Application layer gateway). If you have fixup sip enabled on the firewall (there by default), all packets entering port 5060 is examined and rtp ports are open dynamically as needed. The same is true for trusted calls (from inside interface) and untrusted calls (from outside, dmz interfaces). You will need to perform “conduit permit” commands on the public ip address of Asterisk to allow traffic from untrusted outside interface to come to trusted inside interface on port 5060 with both tcp and udp(all traffic is disabled by default). Please check on the exact syntax of “conduit” permit with cisco docs. I don’t believe you will need to perform this for each RTP port, that should be done automatically by pix ALG. Hope this helps Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Wolfe Sent: Saturday, April 02, 2005 7:03 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] xlite regestration fails but calls to thru While on my network I can register ok with xlite but outside my firewall my Xlite says that regestraion has failed but I am still able to make calls through it. I have opened ports: 5060 udp/tcp and 1-2 udp/tcp is there another port Xlite needs for proper regestration? Is is this a network configuation error on Astrisks part? My Asterisk server is running a IP of 10.0.1.x and my Cisco firewall is passing the public IP address to it from the outside. Thanks for any advice. -Scott ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Discussion Forum
With recent discussions in regards to a forum, I have set-up a multi-faceted Asterisk and Open Source Discussion Board. The link is www.voipnewbie.com/forum It is open and ready for use. Hey Great! Thanks! Just make sure to get linked from the asterisk website (probably in the Digium documentation, and in the user contributed links). I think that's how most new people find this list. Also, make sure you get at least several gurus who are willing to answer questions (I'm guessing you probably are one), but I think there probably needs to be more than one or two. The whole benefit of this list is that 10,000 people read each message coming in (well, skim the subject at least). 5000 of those probably have some facility in asterisk, and hundreds could be qualified as true wizards and gurus. Of course, all those readers and posters can sometimes be detrimental, as was the case in the recent flame war over this subject. - Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New to asterisk.
Hi Paul - I am very new in asterisk community. I just compiled & installed asterisk on a fedora core 3 machine and I want for test purpose to do a small PBX that use X-lite windows sip clients and no trunk for the begining. Where can I find a good how-to to do this job. A small starting how-to that let me understand the principles of setting a PBX with asterisk. The handbook does not like starting guide. There's a great tutorial from OnLamp (the O'Reilly people) to do just what you're looking to do. They recommend using hardphones instead, but the setup is basically the same: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1 With the X-lite clients, just be sure you turn of silence suppression (if you don't it will cause asterisk to hang up on you!). See this: http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite - Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: How does asterisk know the did called on?
Hi Courtney -
If I were to buy 20 did's how do I know within asterisk which number
was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
optimum).
Your provider will "outpulse" a certain number of the DID digits for
you. You may have a choice as to how many digits you want outpulsed.
You can then use these outpulsed digits just as an extension. E.G. if
you have (555) 555-2000 through (555) 555-2019, and you have four
outpulsed digits, you could make them go to your various
extensions/contexts like this:
exten => 2000,1,Goto(IVRMenu,s,1)
exten => 2001,1,Goto(SIPExtensions,101,1)
- Noah
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Re: [Asterisk-Users] Asterisk Voice mail with CCM
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nathan Reeves wrote: | Anyone running Cisco Call Manager and using Asterisk for voice mail | services? Things working well or is the concept a bit of a hassle | to implement? | Hi, I'm using asterisk with a SIP trunk as a voicemail system for CCM without problems till now. João Amaro -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCT/M7JUm/Bor63CERAk+wAJ9oe9EcgbXLERiFBsmfUQv/m23ILACgqqop f/CuLLYESkGmZYuvJzFHA7M= =IaXW -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W
You need to upgrade these phones to the latest firmware for it to work well with asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thore Sent: Sunday, April 03, 2005 3:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi I have a Zyxel P2002 (ATA) with this config. Registration works but i cant call inn. Outgoing works fine. Any clue? Thore - Original Message - From: "Paul Dracevich" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Sunday, April 03, 2005 6:51 AM Subject: RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W > Hi ya I have also three of these phone, here is my entry in my sip.conf > > [4701721] > type=friend > username=4701721 > secret=password721 > host=dynamic > canreinvite=no > context=internal > disallow=all > allow=g729 > dtmfmode=rfc2833 > qualify=4 > permit=0.0.0.0/0.0.0.0 > [EMAIL PROTECTED] > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Ugur > GUNCER > Sent: Sunday, 3 April 2005 4:37 p.m. > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Sip registration Problems With Zyxel P2000W > > Hi all, > > I bougth zyxel wifi phone but i cant register > when i want to register phone to asterisk i recieve > These errors I spend 6 hours to fix regist problem but i cant find the > solution > > [9875] > type=friend > username=9875 > secret=5789 > host=dynamic > context=default > callerid="Ugur Guncer" <9875> > canreinvite=no > dtmfmode=rfc2833 > nat=no > > > > > > > Sip read: > REGISTER sip:213.139.225.82:5060 SIP/2.0 > Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 > From: ;tag=5175B05114E474A31693 > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 12 REGISTER > User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone > Contact: > Expires: 300 > Content-Length: 0 > > > 10 headers, 0 lines > Using latest request as basis request > Sending to 85.99.110.143 : 43956 (non-NAT) > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 > From: ;tag=5175B05114E474A31693 > To: ;tag=as369f8960 > Call-ID: [EMAIL PROTECTED] > CSeq: 12 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Content-Length: 0 > > > to 85.99.110.143:43956 > Transmitting (no NAT): > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 > From: ;tag=5175B05114E474A31693 > To: ;tag=as369f8960 > Call-ID: [EMAIL PROTECTED] > CSeq: 12 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > WWW-Authenticate: Digest realm="asterisk", nonce="0f3403ce" > Content-Length: > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] xlite regestration fails but calls to thru
> While on my network I can register ok with xlite but outside my firewall my > Xlite says that regestraion has failed but I am still able to make calls > through it. I have opened ports: 5060 udp/tcp and 1-2 udp/tcp is > there another port Xlite needs for proper regestration? Is is this a > network configuation error on Astrisks part? My Asterisk server is running a > IP of 10.0.1.x and my Cisco firewall is passing the public IP > address to it from the outside. Registration should occur across udp 5060 only. I don't use a cisco pix, but I believe their is a config command like "sip fixup" (or something like that). Supposedly, the pix will look inside the sip packets and watch for the rtp port negotiation, and then open those udp ports as appropriate. You might check the pix documentation to see exactly how the sip fixup is to be used/defined. In asterisk, you might need "nat=yes" for the external use of xlite. To get more detail as to why the registration is happening correctly, you might want to try "sip debug" and pay attention to IP addresses, error messages, etc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing w/analog phone via FXS port.
> Argh. I can't figure out what I'm doing wrong. I can dial with my SIP > phones just fine, but I want to set up an analog phone plugged into my FXS > port... and, while it gets dialtone, no matter what digit I press, I get > stuff like: > > VERBOSE[21963]: -- Starting simple switch on 'Zap/1-1' > DEBUG[21963]: DTMF digit: 9 on Zap/1-1 > DEBUG[21963]: Hangup: channel: 1 index = 0, normal = 13, callwait = -1, > thirdcall = -1 > DEBUG[21963]: Set option TDD MODE, value: OFF(0) on Zap/1-1 > DEBUG[21963]: Updated conferencing on 1, with 0 conference usersApr 2 > VERBOSE[21963]: -- Hungup 'Zap/1-1' > > I've tried to make it as similar to the SIP stuff in zapata.conf as > possible. Any suggestions on what to read to get this right? I've RTFM'd > no small amount, but, obviously, not the *right* stuff. I'll gladly send > my config files to anyone who wants 'em, or will gladly look at > functioning config files anyone wants to send my way. Without seeing the appropriate sections of your config files, I'd have to take a pure guess that you're not using contexts in the correct way. Can you post just those sections that pertain to this (don't need the entire file)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does asterisk know the did called on?
> If I were to buy 20 did's how do I know within asterisk which number was
> dialed? (like say I want a few of the did's to ring specific extensions
> if they are dialed and others to go through the menu)
>
> Is there any ${var} that has the number dialed in on? (that would be
> optimum).
It varies as to how each provider handles this, but the majority of those
I've tested with send the DID number as the extension dialed.
As an example, when livevoip sends an incoming 800 call (did is the same)
to my * box, they send the call to extn 8001234567 and I handle it in the
extensions.conf like this:
[livevoip800]
exten=>8001234567,1,Dial(SIP/3000,10)
A customer uses a PRI with about 30 did's. Each incoming DID call includes
the dialed number, and the above approach is used to map those incoming
calls to specific extensions.
A variation of the above is to use a GoTo statement (instead of the Dial
statement) to send the call to an existing context/registration.
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Re: [Asterisk-Users] VoIP Provider problems
> No, I'm not ignorant of how this works. You'll notice I put it > "appears" bad when I posted my results. Yes, it's not a perfect way to > show problems -- but taken with a grain of salt it's not half bad. > Especially when sampled over a longer period of time, and if the > original poster can correlate the PingPlotter results to the quality > of his calls. > > Now if he shows 30% loss during good and bad calls, that's another story. > > I posted my results to help the original poster. If he's trying to > troubleshoot an apparent bad connection with Sprint, he needs all the > help he can get. If they can proove the connection works even the > littlest bit, they'll say it's fine and blame Broadvoice. > > If everyone gets similar levels of loss at those points, one could > conclude its a side effect of the routers having better things to do. > But if he's the only one showing them, then it would be a starting > point to conclude something is wrong with his connection or something > along Sprint's backbone. I'm not the original poster either, but for those following this thread keep in mind that a fair number of isp's use an upper-layer device to throttle data flows to some predeteremined rate. For example, I know some cable broadband companies that throttle their users to 128k up and some other value down. Don't have a clue whether their throttling box drops packets, delays them, or what; however, considering they would want to handle both udp and tcp, I'd have to bet some amount they drop udp packets to throttle udp data flows. On the other hand, I know of several dsl broadband companies that don't pay any attention to their uplink congestion, letting their uplink routers drop packets, etc. Since they can't afford to chase uplink utilizations by augmenting bandwidth, dropped packets happen frequently. Nature of the beast for some. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration to multiple GKs
Is it possible to run Asterisk with another GKs using Neighbor mode? If it is possible, we can run asterisk with several gnugks. On Apr 2, 2005 10:41 PM, Alex Vishnev <[EMAIL PROTECTED]> wrote: > I don't think you can. The rules of h323 is so that you can register with a > single gk at a time. > > Alex > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie > Sent: Saturday, April 02, 2005 6:37 AM > To: Asterisk-Users@lists.digium.com > Subject: [Asterisk-Users] Registration to multiple GKs > > Hi all, > > How can I configure chan_h323 or chan_oh323 to register to multiple GK > and route calls in-between? > > Many thanks. > Newbie > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPSwitchboard Version 0.73 Released
Version 0.73 - 3. April 2005. * Italian Language added - Thank you to Francesco Romano for translating * IPSwitchBoard can minimize to tray Download: http://ipswitchboard.thorben.dk IPSwitchBoard is now available in English, Danish and Italian; would you like to help translate IPSwitchBoard? http://ipswitchboard.thorben.dk/index.php?option=com_simpleboard&Itemid=42&f unc=view&id=32&catid=2 Thank you in advance. Thorben ___ IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPSwitchBoard is a FREE Windows.NET application which gives you: * Unattended/attended transfers. * Park calls and retrieve/forward them again. * Organize all your SIP and IAX extensions (automatically retrieved from Asterisk). * Monitor all extensions. * Monitor all queues. * Monitor Agents. * Monitor Parked Calls. * Dynamically log extensions in and out of queues. * Integration with CRM software on the web. * Drop any active call. * Import/Export extensions to/from Asterisk Server DB. * Set Do Not Disturb on Extensions and give a reason. * Speed Dialling. * Share Speed Dial files among all users of IPSwitchBoard. * User selectable ring tones for IPSwitchBoard. * User selectable button colors. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Skinny Call Control Protocol on Asterisk
Hi, I have read in the wiki-pages, that I doesn't need the g729 license, if I use it only in path-thru-mode. Of couse I added AST_FORMAT_G729A to chan_sccp capability, but it dosn't worked. That's why I tried to ask the developer. The Easter version works fine with the * stable, if you add some "#ifdef..." to the header files. There are some definition that you only need with the cvs. But this version is still unstable :-( regards Jens Quoting Remco Barende <[EMAIL PROTECTED]>: I guess if you add the g729 license (or open codec if you are outside the us and don't want to support patents) and add the ability to the driver it should work. Did you see the new version of chan_sccp? The standard Easter version doesn't compile with * stable, the cvs version should. On Sun, 3 Apr 2005, asterisk_on_oelf wrote: Hi, You don't need a licence. Look at: http://chan-sccp.sourceforge.net I use this with a Cisco 7960&7914 and added some of my own patches, but this driver is not stable. It supports only g711 alaw and ulaw, but not g729 (the Cisco-phone does it!) I tried to contact the developer to get and provide some help, but I never got any answer. kind regards Jens Quoting Alexandre Otto Durr <[EMAIL PROTECTED]>: Hi for all! I saw it on http://signate.com/features.php an Open Source PBX Features with support Cisco Skinny Call Control Protocol. Is it possible in Asterisk or I need a license for this? Has anyone using Asterisk with Cisco Skinny? TIA Alexandre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Router with QoS recommendations
Hi List As I have a Cisco PIX 515, with NO QoS functionality, and I’m looking for a router that does outgoing QoS to put in front of my PIX. Problem is that I’m using my 768/8096Kbit ADSL for both data and VoIP, and as soon as data is being sent to the internet the sound quality drops to something that is of NO use. Any suggestions or recommendations is appreciated. Best reg. BennyB ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Discussion Forum
I checked it out. You have indeed created a very functional BBS setup, using open source software. I like it a lot. But you will need to attract a critical mass of Asterisk users in order to succeed in making it an effective Asterisk and VOIP community resource. It takes people to make a BBS work. To pitch in, I have already opened an account on your BBS, and entered my first memo. From the perspective of attracting the necessary critical mass of users, it would have been better if Digium themselves had spearheaded the launching of a proper support BBS, linked from their web page, as the official Asterisk support BBS. However, if a BBS run by an independent will do the trick, then I'm all for it. You have my vote. Tore --- > List: > With recent discussions in regards to a forum, I have set-up a > multi-faceted Asterisk and Open Source Discussion Board. The link is > www.voipnewbie.com/forum It is open and ready for use. > Enjoy! > VoIPNewbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to post my impovements to ASTCC?
You can't see the sweat, but ... I would like tp post my improvements to ASTCC somewhere, ... but where??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Skinny Call Control Protocol on Asterisk
I guess if you add the g729 license (or open codec if you are outside the us and don't want to support patents) and add the ability to the driver it should work. Did you see the new version of chan_sccp? The standard Easter version doesn't compile with * stable, the cvs version should. On Sun, 3 Apr 2005, asterisk_on_oelf wrote: Hi, You don't need a licence. Look at: http://chan-sccp.sourceforge.net I use this with a Cisco 7960&7914 and added some of my own patches, but this driver is not stable. It supports only g711 alaw and ulaw, but not g729 (the Cisco-phone does it!) I tried to contact the developer to get and provide some help, but I never got any answer. kind regards Jens Quoting Alexandre Otto Durr <[EMAIL PROTECTED]>: Hi for all! I saw it on http://signate.com/features.php an Open Source PBX Features with support Cisco Skinny Call Control Protocol. Is it possible in Asterisk or I need a license for this? Has anyone using Asterisk with Cisco Skinny? TIA Alexandre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk.
> Where can I find a good how-to to do this job. A small starting > how-to that let me understand the principles of setting a PBX with > asterisk. The handbook does not like starting guide. Try this: http://automated.it/guidetoasterisk.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Passing varibles *out* of macros
How about setGlobalVar() ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
