Re: [asterisk-users] [FreeBSD 6.3] Zaptel stops responding
On Thu, 19 Jun 2008 11:36:27 +0200, Vincent [EMAIL PROTECTED] wrote: Will do, although it could be a problem in the Zaptel code, which is not written by the mfg. Thanks. I also notice that I can't restart the driver: # /usr/local/etc/rc.d/zaptel restart zaptelkldunload: can't unload file: Device busy zaptelkldload: can't load /usr/local/lib/zaptel/zaptel.ko: File exists ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.21 stalls?
Ip upgraded yesterday from Asterisk 1.4.20.1 to 1.4.21 The update seems to work ok, when asterisk is started all is fine. However after some time it is not possible to call anymore, my Snom display simply shows Not available and incoming calls from the PRI fail, like the PRI is not connected. Reverting back to 1.4.20.1 solves the problem. I tried re-installing and re-compiling 1.4.21 and all the modules several times, didn't help. Anyone else seeing this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXS port doesn't provide dialtone
Hello everyone, I want to connect a fax to an FXS port (TDM420P). For testing purposes, I connected an analogue phone to it first. However, when I pick it up, I cannot hear anything at all. The power cable is plugged into the card, the port is configured to use fxo-signalling. Also, immediate=no. Here's the files: /etc/zaptel.conf: # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: WCTDM/0 Wildcard TDM410P Board 1 # channel 32, WCTDM, no module. # channel 33, WCTDM, no module. fxoks=34 fxoks=35 # Global data loadzone= de defaultzone = de /etc/asterisk/zapata.conf: [trunkgroups] ; define any trunk groups [channels] ; hardware channels ; defaults callreturn=yes callwaiting=yes cancallforward=yes canpark=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes facilityenable=no hidecallerid=no immediate=no language=de overlapdial=yes pridialplan=unknown priindication=outofband prilocaldialplan=unknown resetinterval=3600 restrictcid=no signalling=pri_cpe switchtype=euroisdn threewaycalling=yes transfer=yes usecallerid=yes usecallingpres=yes internationalprefix=+ nationalprefix=0 localprefix=02203 unknownprefix= ; Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) group=1 callgroup=1 pickupgroup=1 context=from-pstn channel = 1-15,17-31 ; Span 2: WCTDM/0 Wildcard TDM410P Board 1 ;;; line=34 WCTDM/0/2 signalling=fxo_ks callerid=Channel 34 6034 group=4 context=from-fax channel = 34 ;;; line=35 WCTDM/0/3 signalling=fxo_ks callerid=Channel 35 6035 group=5 context=from-analog channel = 35 Did I miss anything or configured something wrong? I don't know if any settings for the TE121 interfere with the FXS ports. As a side question, is it possible to 'un-set' any of the directives in some way (so they don't get inherited to the next 'channel =') Kindest regards, Paul ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS port doesn't provide dialtone
On Fri, Jun 20, 2008 at 10:15:48AM +0200, Paul Schewietzek wrote: Hello everyone, I want to connect a fax to an FXS port (TDM420P). For testing purposes, I connected an analogue phone to it first. However, when I pick it up, I cannot hear anything at all. Is Asterisk actually running? Configured to use those channels? What is the output of: cat /proc/zaptel/* asterisk -rx 'zap show channels' -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP development in 1.6 branch?
On Thu, Jun 19, 2008 at 3:50 PM, Paul Belanger [EMAIL PROTECTED] wrote: List, Could anybody speak to the status of development in 1.6 branch? I know support for SIP over TCP is pretty new / experimental but it seems active development of it has slowed or stopped in recent months. Is that a correct statement? Is SIP over TCP more a community project or something headed from Digium? I only ask to get a pulse of its status; not harp or demand people to work on it. Like everybody else, we have some dependencies on SIP over TCP, and have a few bugs open against it. Personally, I would love to help develop or submit patches for the bugs but would need a mentor for that. Either way, just looking to get some more info about the development status of it. I can't speak about the status of SIP/TCP development in Asterisk, but I can say the following: . I've tested Asterisk SIP/TCP and SIP/TLS against a variety of SIP implementations (acting as SIP peers) in a lab setting and things look okay. . I ran into a bug when I register a SIP end-point using SIP/TCP (http://bugs.digium.com/view.php?id=12282). . I think some of the challenges relating to deploying Asterisk SIP/TCP in production environments will be - connection management and NAT traversal. I think certain design thought needs to be put in SIP/TCP feature design to combat these issues. -- Raj Jain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't make asterisk work...how to test?
Most SIP clients have a logging ability.. you can use those.. but turning on debug on the server is the best mechanism, as its whats going on there that counts. sip set debug options And if you want to get really into the lower levels, then tcpdump will let you capture the packets for offline analysis in wireshark. Nmaping against locahost wont tell you much other than an app has 5060 open.. it wont tell you if firewalls are blocking things, or if NAT is an issue. Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of D. Dante Lorenso Sent: 20 June 2008 03:14 To: 'Asterisk Users List' Subject: [asterisk-users] Can't make asterisk work...how to test? All, I've put a new asterisk server at another location and can't seem to get it working. What's the best strategy to debug connections? I'm doing inbound SIP only and have installed the server in the same way as I did on my DEV server. Running an nmap on localhost shows the port listening: -- [asterisk]/ nmap -sU localhost Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2008-06-19 21:12 CDT Interesting ports on localhost.localdomain (127.0.0.1): Not shown: 1476 closed ports PORT STATE SERVICE ... 5060/udp open|filtered sip ... -- [planet]/etc/asterisk nmap -sU localhost Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2008-06-19 20:11 CDT Interesting ports on localhost.localdomain (127.0.0.1): Not shown: 1484 closed ports PORT STATE SERVICE ... 5060/udp open|filtered sip ... -- Is there a command-line tool I can run that will attempt a SIP connection to a SIP server and provide some diagnostics about whether it could authenticate or even connect? -- Dante ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS port doesn't provide dialtone
maggie1:~# cat /proc/zaptel/*
Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) HDB3/CCS/CRC4
IRQ misses: 31
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In use)
6 WCT1/0/6 Clear (In use)
7 WCT1/0/7 Clear (In use)
8 WCT1/0/8 Clear (In use)
9 WCT1/0/9 Clear (In use)
10 WCT1/0/10 Clear (In use)
11 WCT1/0/11 Clear (In use)
12 WCT1/0/12 Clear (In use)
13 WCT1/0/13 Clear (In use)
14 WCT1/0/14 Clear (In use)
15 WCT1/0/15 Clear (In use)
16 WCT1/0/16 HDLCFCS (In use)
17 WCT1/0/17 Clear (In use)
18 WCT1/0/18 Clear (In use)
19 WCT1/0/19 Clear (In use)
20 WCT1/0/20 Clear (In use)
21 WCT1/0/21 Clear (In use)
22 WCT1/0/22 Clear (In use)
23 WCT1/0/23 Clear (In use)
24 WCT1/0/24 Clear (In use)
25 WCT1/0/25 Clear (In use)
26 WCT1/0/26 Clear (In use)
27 WCT1/0/27 Clear (In use)
28 WCT1/0/28 Clear (In use)
29 WCT1/0/29 Clear (In use)
30 WCT1/0/30 Clear (In use)
31 WCT1/0/31 Clear (In use)
Span 2: WCTDM/0 Wildcard TDM410P Board 1
IRQ misses: 5
32 WCTDM/0/0 FXSKS
33 WCTDM/0/1 FXSKS
34 WCTDM/0/2 FXOKS (In use)
35 WCTDM/0/3 FXOKS (In use)
maggie1:~# asterisk -rx 'zap show channels'
Chan Extension Context Language MOH Interpret
pseudofrom-analog de default
1from-pstn de default
2from-pstn de default
3from-pstn de default
4from-pstn de default
5from-pstn de default
6from-pstn de default
7from-pstn de default
8from-pstn de default
9from-pstn de default
10from-pstn de default
11from-pstn de default
12from-pstn de default
13from-pstn de default
14from-pstn de default
15from-pstn de default
17from-pstn de default
18from-pstn de default
19from-pstn de default
20from-pstn de default
21from-pstn de default
22from-pstn de default
23from-pstn de default
24from-pstn de default
25from-pstn de default
26from-pstn de default
27from-pstn de default
28from-pstn de default
29from-pstn de default
30from-pstn de default
31from-pstn de default
34from-faxde default
35from-analog de default
Asterisk is running and working for all SIP phones and the TE121,
connected to an E1 :)
I'm beginning to wonder if the card (the TDM400) is actually OK, or if
the FXS module might be broken...
About the channel configuration: Is it invalid to associate a group to
an analog line? Right now, I said Asterik to Dial(ZAP/g4/${EXTEN}) when
the extension for channel 34 gets called from outside. Could it be
possible that a channel with FXO signalling ignores the group= option in
zapata.conf?
Am Freitag, den 20.06.2008, 11:24 +0300 schrieb Tzafrir Cohen:
On Fri, Jun 20, 2008 at 10:15:48AM +0200, Paul Schewietzek wrote:
Hello everyone,
I want to connect a fax to an FXS port (TDM420P). For testing purposes,
I connected an analogue phone to it first. However, when I pick it up, I
cannot hear anything at all.
Is Asterisk actually running?
Configured to use those channels?
What is the output of:
cat /proc/zaptel/*
asterisk -rx 'zap show channels'
___
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Re: [asterisk-users] GXW 4108 asterisk configuration
Doug wrote: There is a bug in these units that won't let you put punctuation in the extension name. A Grandstream product with a bug... what an unusual concept. cough Seriously, with all the grief I've had with GXP-2000s, BT-200s and GXV-3000s, I wouldn't touch Grandstream gear with a barge pole any more. Yes, the firmware for the GXP-2000s seems to have finally stabilised, but it's taken the better part of three /years/ for this to happen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
Not to my knowledge. The PAP2 is designed as a fairly basic ATA and each line registers as a separate SIP extension. I have no doubt that you could use the call forward feature within the ATA to divert calls to the second extension if the first one was busy, but AFAIK you'd need to register each line separately. Eric Fort wrote: Can the PAP2 be set up such that a second call will ring the second line when the first is busy but only register once with the SIP provider? A beep tone on the same line to denote another incoming call just will not do, The second port needs to act like a seperate line tied to the same DID in a hunt group. Eric On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably looking at the Linksys PAP2 - the functional equivalent of the Sipura SPA-2000. They look different (better if you ask me - the LEDs are far better placed and more useful than they were on the Sipura units) but are pretty much identical under the hood. randulo wrote: On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4858013740251714643830! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4858013740251714643830! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy and delay
Hello, I noticed a bit problematic behavior in ChanSpy function. This is the scenario: 1. agent is making a conversation, 2. I call an extension with ChanSpy and start listening (so far so good), 3. agent completes the call (I am still on the ChanSpy extension), 4. new call is distributed to an agent, 5. agent answers the call (I am still on the ChanSpy extension, but I hear silence...), 6. after 4-5 seconds I can again hear the agent (please see the log: 15:46:38 agent starts the conversation, 4 seconds later chan spy attaches his channel to mine): Jun 19 15:46:38 VERBOSE[9907] logger.c: -- Zap/65-1 answered Local/[EMAIL PROTECTED],2 Jun 19 15:46:42 NOTICE[9905] app_chanspy.c: Attaching Zap/124-1 to Local/[EMAIL PROTECTED],2 Why is there such a delay before I start hearing the agent again? Is this a known bug, is there any workaround to that? I'm using a rather old version of Asterisk: 1.2.9, but I am not thinking about upgrading right now because everything else is working like a charm (and it has to handle more than 3k calls per day). Regards, Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Website callback
Tilghman Lesher [EMAIL PROTECTED] writes: One very big benefit of using a database with cron jobs is that your web application does not need to run as the same user (or otherwise weaken security permissions) as the Asterisk daemon. If running as the same user, you'd have to either set both daemons to the same group (which means the web server has access to all other files that Asterisk writes) or world writable, which is even worse. You can use ACL's. setfacl -m u:webapp:w /path/to/calldir /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [SOLVED] Re: FXS port doesn't provide dialtone
Thanks guys, we solved the problem. There was a non-standart cable between the devices that didn't work correctly... o.O -paul ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXW 4108 asterisk configuration
They still have issues. If you use TCP and reboot the server, the phone will never reconnect as it tries to use a closed TCP session. I opened a ticket with them and after a week their answer is . use udp. Rob Hillis wrote: Doug wrote: There is a bug in these units that won't let you put punctuation in the extension name. A Grandstream product with a bug... what an unusual concept. cough Seriously, with all the grief I've had with GXP-2000s, BT-200s and GXV-3000s, I wouldn't touch Grandstream gear with a barge pole any more. Yes, the firmware for the GXP-2000s seems to have finally stabilised, but it's taken the better part of three /years/ for this to happen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.21 stalls?
Hi Remco, On Fri, 2008-06-20 at 09:33 +0200, Remco Barendse wrote: Ip upgraded yesterday from Asterisk 1.4.20.1 to 1.4.21 I did the same yesterday. The update seems to work ok, when asterisk is started all is fine. Yup. However after some time it is not possible to call anymore, my Snom display simply shows Not available and incoming calls from the PRI fail, like the PRI is not connected. That's not good at all. Reverting back to 1.4.20.1 solves the problem. I tried re-installing and re-compiling 1.4.21 and all the modules several times, didn't help. Anyone else seeing this? Nope but on this 1.4.21 box I am using an Eicon Diva Server card and that has always worked rock solid. Make sure you start Asterisk with -g so it can dump core, set appropriate values for ulimit and log everything (logger.conf). Maybe you will then get more info what's going on. Good luck! Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] commercial discussion
Hey can you guys give me a mailing list/group where I can have little on commercial discussion? Regards jimmy --- On Thu, 6/19/08, Steve Totaro [EMAIL PROTECTED] wrote: From: Steve Totaro [EMAIL PROTECTED] Subject: Re: [asterisk-users] Trouble with PRI config To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, June 19, 2008, 5:40 PM pri_net usually when connecting to a legacy system. Thanks, Steve T On Thu, Jun 19, 2008 at 1:38 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.21 stalls?
Remco Barendse pisze: However after some time it is not possible to call anymore, my Snom display simply shows Not available and incoming calls from the PRI fail, like the PRI is not connected. Reverting back to 1.4.20.1 solves the problem. I tried re-installing and re-compiling 1.4.21 and all the modules several times, didn't help. Anyone else seeing this? I have same problem. I'm just downgrading to 1.4.20.1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream Busy Light Fields
Hello Gordon,
On Thu, 19 Jun 2008, Jan Prunk wrote:
* You might want to try:
**
** exten = _**.,1,Pickup(${EXTEN:2})
** exten = _**.,n,Hangup()
*
*
** Ok I have tried adding these 2 lines, and the error which I get when calling
** 01 5863165, which then rings extension 65, and I try to accept the call on
** extension 70 by a BLF button. It gives me error code.
**
** -- Accepting overlap voice call from '015852977' to '5863165' on channel
** 0/1, span 3
**-- Starting simple switch on 'Zap/7-1'
**-- Executing [5863165 at buster
http://lists.digium.com/mailman/listinfo/asterisk-users:1]
Dial(Zap/7-1, SIP/65|17|rtk) in new
** stack
** Extension Changed 65[BLF] new state Ringing for Notify User 70
**-- Called 65
**-- SIP/65-081fb370 is ringing
**-- Executing [**65 at buster
http://lists.digium.com/mailman/listinfo/asterisk-users:1]
PickUp(SIP/70-b5f18268, 65) in new
** stack
** [2008-06-19 15:13:33] WARNING[7287]: channel.c:4347 ast_get_group: Ignoring
** invalid group 65 (maximum group is 63)
**-- No channel found 0.
** == Spawn extension (buster, **65, 1) exited non-zero on 'SIP/70-b5f18268'
**-- Channel 0/1, span 3 got hangup request, cause 16
** Extension Changed 65[BLF] new state Idle for Notify User 70
** == Spawn extension (buster, 5863165, 1) exited non-zero on 'Zap/7-1'
**-- Hungup 'Zap/7-1'
*
Er, I don't get quite the same output as you - I'm on 1.2 though. A test
call I've just done - extension 109 called extension 100, and extension
101 (a grandstream phone) picked it up by pushing the BLF key
corresponding to extension 100:
-- Executing Dial(SIP/109-0820a178, IAX2/100SIP/100||WwTton)
in new stack
-- Called 100
-- SIP/100-081fe780 is ringing
Extension Changed 100 new state Ringing for Notify User 101
-- Executing Pickup(SIP/101-081edf38, 100) in new stack
-- Executing Hangup(SIP/101-081edf38, ) in new stack
== Spawn extension (internal, **100, 2) exited non-zero on 'SIP/101-081edf38'
-- SIP/101-081edf38 answered SIP/109-0820a178
Extension Changed 100 new state Idle for Notify User 101
== Spawn extension (macro-dialInternal, s, 53) exited non-zero on
'SIP/109-0820a178' in macro 'dialInternal'
So your pickup is picking up a group - seems odd to me, but maybe the
behaviour changed after 1.2 ?
One other thing - do you have
exten = 65,1,Dial(SIP/65)
Yes I do have this:
exten = 65,1,Dial(SIP/65,30,rtk)
exten = 65,n,Hangup()
exten = 70,1,Dial(SIP/70,30,rtk)
exten = 70,n,Hangup()
As pickup works on the extension not the channel... (ie. what do you dial
on a phone to make the SIP/65 ring? What does the DDI point to?)
It goes like this:
exten = 5863165,1,Dial(SIP/65,17,rtk)
exten = 5863165,n,Dial(SIP/64,120,rtk)
exten = 5863165,n,Hangup()
Regards,
Jan
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Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86
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Re: [asterisk-users] commercial discussion
You can ask any hardware/soft choice/cost here, but for support and commearcial offers, asterisk-biz or asterisk-biz forum on asterisk.org 2008/6/20 Jimmy Jill [EMAIL PROTECTED]: Hey can you guys give me a mailing list/group where I can have little on commercial discussion? Regards jimmy --- On *Thu, 6/19/08, Steve Totaro [EMAIL PROTECTED]*wrote: From: Steve Totaro [EMAIL PROTECTED] Subject: Re: [asterisk-users] Trouble with PRI config To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, June 19, 2008, 5:40 PM pri_net usually when connecting to a legacy system. Thanks, Steve T On Thu, Jun 19, 2008 at 1:38 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Busy Light Fields
On Fri, 20 Jun 2008, Jan Prunk wrote: One other thing - do you have exten = 65,1,Dial(SIP/65) Yes I do have this: exten = 65,1,Dial(SIP/65,30,rtk) exten = 65,n,Hangup() exten = 70,1,Dial(SIP/70,30,rtk) exten = 70,n,Hangup() As pickup works on the extension not the channel... (ie. what do you dial on a phone to make the SIP/65 ring? What does the DDI point to?) It goes like this: exten = 5863165,1,Dial(SIP/65,17,rtk) exten = 5863165,n,Dial(SIP/64,120,rtk) exten = 5863165,n,Hangup() Right. That's the issue. You'll need to pickup 5863165, not 65. (or 64). Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR for callee (called party)
Thanks Tony,
First of all, thanks for answer.
The possible solution to solve the problem with auto hangup is to use 'h'
extension, which can execute some commands after hanging up, here we call
MeetMeAdmin(confno,K) from either caller or callee, what will hang up call
when caller drops the call or callee.
Actually not the best solution.
Something like that:
*
[Prompt]
exten = s,1,Goto(40)
exten = s,2,Playback(hello1)
exten = s,n,MeetMe(confno|qx)
exten = s,n,Hangup()
exten = s,40,Playback(hello2)
exten = s,n,MeetMe(confno|qx)
exten = s,n,Hangup()
exten = h,1,MeetMeAdmin(confno,K)
[Main]
.Dial(...G(Prompt^s^1)*
On Thu, Jun 19, 2008 at 6:26 PM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Alexander Olekhnovich [EMAIL PROTECTED] wrote:
I'm trying to make the next scenario in Asterisk DialPlan: Alice calls
Bob,
Asterisk executes Dial application with G(context^exten^pri), after that
Bob
answers the call, Asterisk transfers Alice to pri, Bob to pri+1. It
should
be possible for example that in that context Asterisk executes different
scenarios for Bob and Alice and then connects Alice to Bob to let them
communicate. The problem is that I can not connect both sides for
conversation, Asterisk just hangs up after executes the scenarios.
*[AnswerPrompt]
exten = s,1,Goto(10)
exten = s,2,Playback(Announce1)
exten = s,10,Playback(Announce2)
[call-number]
exten = _X.,1,Dial(SIP/${EXTEN}|G(AnswerPrompt^s^1))
exten = _X.,n,Hangup()
*
Is there any solutions? Any help will be appropriate.
In most versions of Asterisk, the best you can do is to put both calls
into a Meetme room with a unique room number. The drawback with that is
that when one of the parties hangs up, it doesn't automatically hang up
the other party.
There have been one or two enhancements proposed in the past to allow
one channel to grab another and bridge to it, but I don't think such an
application has made it into official versions yet (1.4 or trunk).
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Best Regards
Alexander Olekhnovich
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Re: [asterisk-users] ChanSpy and delay
I will answer my own question :-) The following line in the app_chanspy.c was causing my problem: waitms = count ? 100 : 5000; and of course sequentially then the line: res = ast_waitfordigit(chan, waitms); in the same beginning of the same for loop but in the next iteration. I have changed waitms = count ? 100 : 5000; to waitms = count ? 100 : 500; and I didn't notice any change (besides my problem being fixed, which is good :-) ), especially no negative effects. I will however still test this thing, but maybe some of the development guys might tell me what exactly have I changed and what side-effects it might cause :-) ? With Regards, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Friday, June 20, 2008 12:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] ChanSpy and delay Hello, I noticed a bit problematic behavior in ChanSpy function. This is the scenario: 1. agent is making a conversation, 2. I call an extension with ChanSpy and start listening (so far so good), 3. agent completes the call (I am still on the ChanSpy extension), 4. new call is distributed to an agent, 5. agent answers the call (I am still on the ChanSpy extension, but I hear silence...), 6. after 4-5 seconds I can again hear the agent (please see the log: 15:46:38 agent starts the conversation, 4 seconds later chan spy attaches his channel to mine): Jun 19 15:46:38 VERBOSE[9907] logger.c: -- Zap/65-1 answered Local/[EMAIL PROTECTED],2 Jun 19 15:46:42 NOTICE[9905] app_chanspy.c: Attaching Zap/124-1 to Local/[EMAIL PROTECTED],2 Why is there such a delay before I start hearing the agent again? Is this a known bug, is there any workaround to that? I'm using a rather old version of Asterisk: 1.2.9, but I am not thinking about upgrading right now because everything else is working like a charm (and it has to handle more than 3k calls per day). Regards, Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Busy Light Fields
Hello Gordon, Same error if I change the extension ...to ring directly 5863165 (see the maximum group number is 63) -- Accepting overlap voice call from '015852977' to '5863165' on channel 0/1, span 1 -- Starting simple switch on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, SIP/5863165|17|rtk) in new stack Extension Changed 5863165[BLF] new state Ringing for Notify User 70 -- Called 5863165 -- SIP/5863165-08245258 is ringing Extension Changed 70[BLF] new state InUse for Notify User 65 Extension Changed 70[BLF] new state InUse for Notify User 60 -- Executing [EMAIL PROTECTED]:1] PickUp(SIP/70-b5f1fef8, [EMAIL PROTECTED]) in new stack [2008-06-20 15:38:19] WARNING[9772]: channel.c:4347 ast_get_group: Ignoring invalid group 5863165 (maximum group is 63) -- No channel found 0. == Spawn extension (buster, **5863165, 1) exited non-zero on 'SIP/70-b5f1fef8' Extension Changed 70[BLF] new state Idle for Notify User 65 (queued) Extension Changed 70[BLF] new state Idle for Notify User 60 (queued) -- Channel 0/1, span 1 got hangup request, cause 16 Extension Changed 5863165[BLF] new state Idle for Notify User 70 == Spawn extension (buster, 5863165, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Regards, Jan On Fri, Jun 20, 2008 at 3:01 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Fri, 20 Jun 2008, Jan Prunk wrote: One other thing - do you have exten = 65,1,Dial(SIP/65) Yes I do have this: exten = 65,1,Dial(SIP/65,30,rtk) exten = 65,n,Hangup() exten = 70,1,Dial(SIP/70,30,rtk) exten = 70,n,Hangup() As pickup works on the extension not the channel... (ie. what do you dial on a phone to make the SIP/65 ring? What does the DDI point to?) It goes like this: exten = 5863165,1,Dial(SIP/65,17,rtk) exten = 5863165,n,Dial(SIP/64,120,rtk) exten = 5863165,n,Hangup() Right. That's the issue. You'll need to pickup 5863165, not 65. (or 64). Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jan Prunk janprunk AT SPAMFREE gmail DOT com Website: http://www.prunk.si PGP key: 00E80E86 Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS port doesn't provide dialtone
Hi not the issue here, but yo asked and thus I'll answer: On Fri, Jun 20, 2008 at 11:51:27AM +0200, Paul Schewietzek wrote: Could it be possible that a channel with FXO signalling ignores the group= option in zapata.conf? A. no problem with that. B. This is only related to dialing out, not to incoming calls. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.21 stalls?
On Friday 20 June 2008 02:33:04 Remco Barendse wrote: Ip upgraded yesterday from Asterisk 1.4.20.1 to 1.4.21 The update seems to work ok, when asterisk is started all is fine. However after some time it is not possible to call anymore, my Snom display simply shows Not available and incoming calls from the PRI fail, like the PRI is not connected. Reverting back to 1.4.20.1 solves the problem. I tried re-installing and re-compiling 1.4.21 and all the modules several times, didn't help. Anyone else seeing this? If you could recompile with DONT_OPTIMIZE and DEBUG_THREADS enabled in the compiler flags (in 'make menuselect') and obtain a 'core show locks' when this occurs, then upload the resulting output to an issue on bugs.digium.com, that would help a great deal in tracking down where exactly the problem lies. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Busy Light Fields
Hi,
[2008-06-20 15:38:19] WARNING[9772]: channel.c:4347 ast_get_group: Ignoring
invalid group 5863165 (maximum group is 63)
we had a similar error, found somewhere (voip-info.org?) this solution:
exten = _**2XX,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten = _**2XX,n,PickUp(${EXTEN:2})
where 2XX are our internals...
don't remember why, but it works with GS GXP2010s and with
Thomson ST2030s.
Cheers
--
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[EMAIL PROTECTED] ..o () ascii ribbon campaign
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Re: [asterisk-users] GXW 4108 asterisk configuration
Like I said, I wouldn't touch Grandstream gear with a barge pole any more. I can point you at at a few companies that hold exactly the same opinion too. Don't expect any useful action on the TCP stuff anytime soon - at least not on a version of firmware that doesn't introduce any other nasty problems. This is one reason I won't upgrade from 1.1.6.16 for a very long time on the one install I have to support with Grandstreams (since I recommended them in the first place) This is working nicely here, and I've been bitten too many times by upgrading to a new improved version that you subsequently can't downgrade from when you discover some of the undocumented improvements. Peder @ NetworkOblivion wrote: They still have issues. If you use TCP and reboot the server, the phone will never reconnect as it tries to use a closed TCP session. I opened a ticket with them and after a week their answer is . use udp. Rob Hillis wrote: Doug wrote: There is a bug in these units that won't let you put punctuation in the extension name. A Grandstream product with a bug... what an unusual concept. cough Seriously, with all the grief I've had with GXP-2000s, BT-200s and GXV-3000s, I wouldn't touch Grandstream gear with a barge pole any more. Yes, the firmware for the GXP-2000s seems to have finally stabilised, but it's taken the better part of three /years/ for this to happen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:485b955a40254742053641! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't make asterisk work...how to test?
All, I did finally get my server up and running. I thought I'd share some tools I used. 1) use this to see if asterisk is even listening on port 5060 with UDP nmap -sU -p5058-5062 localhost Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2008-06-20 10:49 CDT Interesting ports on localhost.localdomain (127.0.0.1): PORT STATE SERVICE 5058/udp closedunknown 5059/udp closedunknown 5060/udp open|filtered sip 5061/udp closedunknown 5062/udp closedunknown Nmap finished: 1 IP address (1 host up) scanned in 1.224 seconds 2) use TCPdump to see of any traffic is going through the sip port tcpdump -v -pi eth0 udp port 5060 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes 11:55:39.155542 IP (tos 0x0, ttl 51, id 8710, offset 0, flags [DF], proto: UDP (17), length: 779) 100.ipcomms.net.sip asterisk.larkspark.dev.sip: SIP, length: 751 INVITE sip:[EMAIL PROTECTED] SIP/2.0 3) turn on sip debugging in asterisk asterisk -cvv asterisk*CLI sip set debug Eventually I did find a few problems and was able to correct them. It's pretty much just what Adrian suggested. -- Dante Adrian Marsh wrote: Most SIP clients have a logging ability.. you can use those.. but turning on debug on the server is the best mechanism, as its whats going on there that counts. sip set debug options And if you want to get really into the lower levels, then tcpdump will let you capture the packets for offline analysis in wireshark. Nmaping against locahost wont tell you much other than an app has 5060 open.. it wont tell you if firewalls are blocking things, or if NAT is an issue. Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of D. Dante Lorenso Sent: 20 June 2008 03:14 To: 'Asterisk Users List' Subject: [asterisk-users] Can't make asterisk work...how to test? All, I've put a new asterisk server at another location and can't seem to get it working. What's the best strategy to debug connections? I'm doing inbound SIP only and have installed the server in the same way as I did on my DEV server. Running an nmap on localhost shows the port listening: -- [asterisk]/ nmap -sU localhost Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2008-06-19 21:12 CDT Interesting ports on localhost.localdomain (127.0.0.1): Not shown: 1476 closed ports PORT STATE SERVICE ... 5060/udp open|filtered sip ... -- [planet]/etc/asterisk nmap -sU localhost Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2008-06-19 20:11 CDT Interesting ports on localhost.localdomain (127.0.0.1): Not shown: 1484 closed ports PORT STATE SERVICE ... 5060/udp open|filtered sip ... -- Is there a command-line tool I can run that will attempt a SIP connection to a SIP server and provide some diagnostics about whether it could authenticate or even connect? -- Dante ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming calls through callcentric sip account!!
Hi all, i've recently acquired a callcentric account. i've perfectly setup my sip.conf and extensions.conf to make outgoing calls. but the problem is with incoming calls! when i call in, asterisk doesnt even see the incoming call! how is tht possible! please see the following my config: sip.conf: [general] dtmfmode = rfc2833 context=from-callcentric srvlookup=yes register = username:[EMAIL PROTECTED]/username [callcentric] type=peer context=from-callcentric host=callcentric.com username=username secret=password fromuser=username fromdomain=callcentric.com insecure=very [107] context=to-callcentric type=friend username=107 secret=secret host=dynamic etensions.conf: [from-callcentric] exten = s,1,Dial(SIP/107) [to-callcentric] exten = _0.,1,Dial(SIP/[EMAIL PROTECTED]) _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Website callback
Great, thanks guys! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 19, 2008 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Website callback On Thu, Jun 19, 2008 at 9:57 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jun 19, 2008 at 08:05:59AM -0500, Tilghman Lesher wrote: On Thursday 19 June 2008 07:57:07 Mark Hamilton wrote: LOL, I agree, it _did_ sound a little complicated than to just schedule a call in the future. I apologize for not being able to find this on the wiki earlier when I searched. The other cron jobs and everything probably bring _something_ to the table. I wonder what. Either way, please keep 'em coming boys, and yes I'd like to know the answer to Tzafrir's question about performance. Test it yourself? for i in `seq 1500`; do something to create a call file sleep a_bit done One very big benefit of using a database with cron jobs is that your web application does not need to run as the same user (or otherwise weaken security permissions) as the Asterisk daemon. If running as the same user, you'd have to either set both daemons to the same group (which means the web server has access to all other files that Asterisk writes) or world writable, which is even worse. In any version you'll still need something with permissions to originate calls on Asterisk. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir I have done hundreds at once, takes a few seconds to handle (all SIP) Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation
Hi All, I've been playing with Openfire Asterisk-IM plugin installed on the same server with Asterisk 1.4 with MySQL as the Openfire database. Using Spark IM as the client on user machines. It seems to work fairly well, not too bad to install. This first thing I notice is all the packages that must be installed to get Openfire running, java-jre, mysql (needed for asterisk-im to work) and many other dependencies. So now the PBX is over 1.2 Gig for the installation. Typical PBX installs are under 600 Meg. This makes me wonder about server stability, reliability and performance as uptime creeps on and user count increases over 50 to 100+. Can anyone give me feedback on real world experience with this type of setup and any performance issues that my arise? Is it better for production to run Openfire on a separate server than the PBX? My biggest concern is deploying a 100+ user environment with high call volume and high chat volume. Java seems to be a bit resource hungry with the user notifications and call pop ups. I would hate to have the IM server walking over Asterisk and affecting call quality or PBX stability. Thanks. JR - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation
On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson [EMAIL PROTECTED] wrote: So now the PBX is over 1.2 Gig for the installation. Typical PBX installs are under 600 Meg. This makes me wonder about server stability, reliability and performance as uptime creeps on and user count increases over 50 to 100+. Increased data on the hard drive won't really have an affect on reliability or performance. Can anyone give me feedback on real world experience with this type of setup and any performance issues that my arise? I can't speak directly to the asterisk + openfire situation. I can, however, say that I've been running openfire for nearly a year now on a very highly-loaded server (other than openfire, it's running nagios and cacti, monitoring about 300 devices around our network) - the load average on this 5-year single processor old dell server is pegged near 1.00 24x7. I haven't had a single problem with openfire, and I have between 50 and 100 open sessions at any one time. In the year that I've been running openfire, I've only had to restart it once, and that was to upgrade the software. It takes very little CPU, and a modest amount of RAM. Is it better for production to run Openfire on a separate server than the PBX? What's your definition of better. Is it better to not have all your eggs in one basket? Is it better to only need to purchase one server? Is it better to only have one server to manage/update/etc versus two? My biggest concern is deploying a 100+ user environment with high call volume and high chat volume. Java seems to be a bit resource hungry with the user notifications and call pop ups. I would hate to have the IM server walking over Asterisk and affecting call quality or PBX stability. Speaking personally, I'd have no problems putting openfire and asterisk on the same box. If needed, you could even just nice the openfire process down to a lower priority than asterisk - it's not as latency-sensitive as asterisk is. I'd doubt you'll need to do that, though. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.
If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very
extensions.conf
[my-phones]
exten = 2000,1,Dial(SIP/2000)
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 2003,1,Dial(SIP/2003)
exten = 6000,1,MeetMe(600,i,54321)
;include = lan-phones
[bogon-calls]
exten = _.,1,Congestion
[pstn-incoming]
include = lan-phones
[local-phones]
include = lan-phones
include = pstn-outbound
[pstn-outbound]
; Calls starting with 9 have the 9 stripped are then routed out to the
PSTN
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten = _9,1,Dial,SIP/[EMAIL PROTECTED] ; IP address of Cisco
gateway
;exten = _9,2,Congestion
exten = _9.,2,Congestion
[lan-phones]
exten = 1001,1,Dial(SIP/1001,20)
exten = 1001,2,Voicemail(u1001)
exten = 1001,3,Answer(SIP/1001)
exten = 1001,102,Voicemail(b1001)
exten = 1001,103,Hangup
Cisco 2611 config
Building configuration...
Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1...
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx:5060
session transport udp
dtmf-relay h245-alphanumeric
clid strip
no vad
!
dial-peer voice 1 pots
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:xxx.xxx.xxx.xxx
!
!
!
telephony-service
transfer-pattern
transfer-system full-blind
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
password
login
!
!
end
Thank you
Sang-Kil (Sam) Suh
System administrator
--
Ticoon Technology Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and remote phone.
Hi everyone! I have been reading for a couple of days online in order to setup a remote phone, but no luck so far. The remote phone registers and I can call extensions, but the call only lasts 19-21 seconds before it drops (when I call from remote phone). When someone calls the remote phone we cannot hear each other. The scenario is de following: Office -Router configuration: -Ports being forwarded to Asterisk -UDP = 5060 -UDP = 5004 -UDP = 5060-5065 (configured range on rpc.conf first) House -Westell Modem configured as IP-Passthrough. In other words, the phone has a static public IP address. Phone: -STUN Server configured. I also had to configured symmetrical RTP because on the other side they couldn't hear me. The call only lasts 18-21 seconds until it drops. Asterisk Configuration Files: -I configured external and lan IP. -Enabled NAT for the remote extension. What am I missing? Why is it that I can call from remote phone and hear the person, but they cannot hear me. When the office calls the house we cannot hear each other. Today I configured Asterisk with a public IP address and the home phone work just fine. However, I do think think this is the best setup because having Asterisk expose to the internet is not a good idea. I want to be able to have remote phones with Asterisk behind a router (private IP address). Thanks in advance. Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com Fidel Garcia System Engineer sysTeam. 7205 NW 19th Street, Suite 302 Miami, Florida 33126 Email: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice only works from one way.
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.
If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very
extensions.conf
[my-phones]
exten = 2000,1,Dial(SIP/2000)
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 2003,1,Dial(SIP/2003)
exten = 6000,1,MeetMe(600,i,54321)
;include = lan-phones
[bogon-calls]
exten = _.,1,Congestion
[pstn-incoming]
include = lan-phones
[local-phones]
include = lan-phones
include = pstn-outbound
[pstn-outbound]
; Calls starting with 9 have the 9 stripped are then routed out to the
PSTN
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten = _9,1,Dial,SIP/[EMAIL PROTECTED] ; IP address of Cisco
gateway
;exten = _9,2,Congestion
exten = _9.,2,Congestion
[lan-phones]
exten = 1001,1,Dial(SIP/1001,20)
exten = 1001,2,Voicemail(u1001)
exten = 1001,3,Answer(SIP/1001)
exten = 1001,102,Voicemail(b1001)
exten = 1001,103,Hangup
Cisco 2611 config
Building configuration...
Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1...
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx:5060
session transport udp
dtmf-relay h245-alphanumeric
clid strip
no vad
!
dial-peer voice 1 pots
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:xxx.xxx.xxx.xxx
!
!
!
telephony-service
transfer-pattern
transfer-system full-blind
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
password
login
!
!
end
Thank you
Sang-Kil (Sam) Suh
System administrator
--
Ticoon Technology Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To
Re: [asterisk-users] xxxx SPAM Low xxxx Re: Voice only works from one way.
Yes. Both Asterisk and Cisco is behind it.
On 6/20/08 3:26 PM, Sam Tam [EMAIL PROTECTED] wrote:
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.
If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very
extensions.conf
[my-phones]
exten = 2000,1,Dial(SIP/2000)
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 2003,1,Dial(SIP/2003)
exten = 6000,1,MeetMe(600,i,54321)
;include = lan-phones
[bogon-calls]
exten = _.,1,Congestion
[pstn-incoming]
include = lan-phones
[local-phones]
include = lan-phones
include = pstn-outbound
[pstn-outbound]
; Calls starting with 9 have the 9 stripped are then routed out to the
PSTN
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten = _9,1,Dial,SIP/[EMAIL PROTECTED] ; IP address of Cisco
gateway
;exten = _9,2,Congestion
exten = _9.,2,Congestion
[lan-phones]
exten = 1001,1,Dial(SIP/1001,20)
exten = 1001,2,Voicemail(u1001)
exten = 1001,3,Answer(SIP/1001)
exten = 1001,102,Voicemail(b1001)
exten = 1001,103,Hangup
Cisco 2611 config
Building configuration...
Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1...
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx:5060
session transport udp
dtmf-relay h245-alphanumeric
clid strip
no vad
!
dial-peer voice 1 pots
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:xxx.xxx.xxx.xxx
!
!
!
telephony-service
transfer-pattern
transfer-system full-blind
!
!
line con 0
exec-timeout 0 0
line aux 0
Re: [asterisk-users] Voice only works from one way.
Yes, both Asterisk and Cisco are behind Nat.
On 6/20/08 3:26 PM, Sam Tam [EMAIL PROTECTED] wrote:
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.
If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very
extensions.conf
[my-phones]
exten = 2000,1,Dial(SIP/2000)
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 2003,1,Dial(SIP/2003)
exten = 6000,1,MeetMe(600,i,54321)
;include = lan-phones
[bogon-calls]
exten = _.,1,Congestion
[pstn-incoming]
include = lan-phones
[local-phones]
include = lan-phones
include = pstn-outbound
[pstn-outbound]
; Calls starting with 9 have the 9 stripped are then routed out to the
PSTN
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten = _9,1,Dial,SIP/[EMAIL PROTECTED] ; IP address of Cisco
gateway
;exten = _9,2,Congestion
exten = _9.,2,Congestion
[lan-phones]
exten = 1001,1,Dial(SIP/1001,20)
exten = 1001,2,Voicemail(u1001)
exten = 1001,3,Answer(SIP/1001)
exten = 1001,102,Voicemail(b1001)
exten = 1001,103,Hangup
Cisco 2611 config
Building configuration...
Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1...
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx:5060
session transport udp
dtmf-relay h245-alphanumeric
clid strip
no vad
!
dial-peer voice 1 pots
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:xxx.xxx.xxx.xxx
!
!
!
telephony-service
transfer-pattern
transfer-system full-blind
!
!
line con 0
exec-timeout 0 0
line aux 0
[asterisk-users] Calls Disconnect very often
Greetings.. i'm having a disconnection problems with Calls comming to my Call Center.. i'm using the free version of G729 .. and i'm starting to suspect it would be the reason.. i just need to know if it's possible or there will be other problems?? i asked the technician in the location to do a ping to our DNS and there seems to be a problem.. one of them doesn't response and the other one has lots of timeouts.. he noticed that the disconnection of the call occures at the same time a timeout happens to the DNS .. so my question will be the following.. is it a CODEC problem? G729 more sensitive to internet than G723? Regards Tark Sawah _ The i’m Talkathon starts 6/24/08. For now, give amongst yourselves. http://www.imtalkathon.com?source=TXT_EML_WLH_LearnMore_GiveAmongst___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Disconnect very often
Sounds like a network connectivity issue. Start at your physical layer and work up. Tariq .. wrote: Greetings.. i'm having a disconnection problems with Calls comming to my Call Center.. i'm using the free version of G729 .. and i'm starting to suspect it would be the reason.. i just need to know if it's possible or there will be other problems?? i asked the technician in the location to do a ping to our DNS and there seems to be a problem.. one of them doesn't response and the other one has lots of timeouts.. he noticed that the disconnection of the call occures at the same time a timeout happens to the DNS .. so my question will be the following.. is it a CODEC problem? G729 more sensitive to internet than G723? Regards Tark Sawah -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation
See below: Erik Anderson wrote: On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson [EMAIL PROTECTED] wrote: So now the PBX is over 1.2 Gig for the installation. Typical PBX installs are under 600 Meg. This makes me wonder about server stability, reliability and performance as uptime creeps on and user count increases over 50 to 100+. Increased data on the hard drive won't really have an affect on reliability or performance. Can anyone give me feedback on real world experience with this type of setup and any performance issues that my arise? I can't speak directly to the asterisk + openfire situation. I can, however, say that I've been running openfire for nearly a year now on a very highly-loaded server (other than openfire, it's running nagios and cacti, monitoring about 300 devices around our network) - the load average on this 5-year single processor old dell server is pegged near 1.00 24x7. I haven't had a single problem with openfire, and I have between 50 and 100 open sessions at any one time. In the year that I've been running openfire, I've only had to restart it once, and that was to upgrade the software. It takes very little CPU, and a modest amount of RAM. Is it better for production to run Openfire on a separate server than the PBX? What's your definition of better. Is it better to not have all your eggs in one basket? Is it better to only need to purchase one server? Is it better to only have one server to manage/update/etc versus two? My biggest concern is deploying a 100+ user environment with high call volume and high chat volume. Java seems to be a bit resource hungry with the user notifications and call pop ups. I would hate to have the IM server walking over Asterisk and affecting call quality or PBX stability. Speaking personally, I'd have no problems putting openfire and asterisk on the same box. If needed, you could even just nice the We run with the openfire process on the same box as the * server - we have not had a single problem with openfire in over 2 years now. openfire process down to a lower priority than asterisk - it's not as latency-sensitive as asterisk is. I'd doubt you'll need to do that, though. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommendations for Motel Instalation.
Dear List, I have a customer who owns a little Motel, and he wants to upgrade to a Asterisk PBX. There is one analog phone per room (aprox 80), and the cable is CAT 3. Any recommendations on what card to use? TDM24XXP vs Channel Bank? Regards, Ing. Arturo Ochoa N Electrosystems S RL Tel. (656)-6230794 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice only works from one way.
I was never able to get it to work that way. When I had Asterisk in NAT I
was able to make calls, but most of the times they were one way voice.
I was able to get two-way voice when I configured the remote phone using
STUN and Symetrical RTP. However, the calls dropped every 19-20 seconds. I
read several threads online, but nobody explained the requirements in
details. Everything works fine if you have a public IP address or DMZ on
Asterisk.
Good luck and please let me know if you get it up and running.
Fidel Garcia
System Engineer
sysTeam.
7205 NW 19th Street, Suite 302
Miami, Florida 33126
Email: [EMAIL PROTECTED]
Tel: (305)-477-7303 Fax: (305)-477-0013
http://www.systeamusa.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Friday, June 20, 2008 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice only works from one way.
Yes, both Asterisk and Cisco are behind Nat.
On 6/20/08 3:26 PM, Sam Tam [EMAIL PROTECTED] wrote:
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.
If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very
extensions.conf
[my-phones]
exten = 2000,1,Dial(SIP/2000)
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 2003,1,Dial(SIP/2003)
exten = 6000,1,MeetMe(600,i,54321)
;include = lan-phones
[bogon-calls]
exten = _.,1,Congestion
[pstn-incoming]
include = lan-phones
[local-phones]
include = lan-phones
include = pstn-outbound
[pstn-outbound]
; Calls starting with 9 have the 9 stripped are then routed out to the
PSTN
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten = _9,1,Dial,SIP/[EMAIL PROTECTED] ; IP address of Cisco
gateway
;exten = _9,2,Congestion
exten = _9.,2,Congestion
[lan-phones]
exten = 1001,1,Dial(SIP/1001,20)
exten = 1001,2,Voicemail(u1001)
exten = 1001,3,Answer(SIP/1001)
exten = 1001,102,Voicemail(b1001)
exten = 1001,103,Hangup
Cisco 2611 config
Building configuration...
Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
Re: [asterisk-users] Recommendations for Motel Instalation.
80 rooms? I guess you and I have slightly differing opinions as to what a small motel is. :) If you have 80 analogue channels, then you'd need 4 TDM2400P cards. Unless your server is powered by a small nuclear reactor, you'll be better off with either 3 E1 or 4 T1 channels banks or 2 32 port XR0008 and 1 16 port XR0006 Astribanks. Arturo Ochoa wrote: Dear List, I have a customer who owns a little Motel, and he wants to upgrade to a Asterisk PBX. There is one analog phone per room (aprox 80), and the cable is CAT 3. Any recommendations on what card to use? TDM24XXP vs Channel Bank? Regards, Ing. Arturo Ochoa N Electrosystems S RL Tel. (656)-6230794 !DSPAM:485c27f840251159815309! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:485c27f840251159815309! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice only works from one way.
Well to be honest, our experience with asterisk never works with under NAT.
if you got DMZ then it will otherwise don't hold your breath for it.
If you want to use it as a production server
Your option is 1. Get a Real IP
2. there is no 2 really just get an ReaL Public IP
Sam
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fidel Garcia
Sent: Saturday, June 21, 2008 6:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Voice only works from one way.
I was never able to get it to work that way. When I had Asterisk in NAT I
was able to make calls, but most of the times they were one way voice.
I was able to get two-way voice when I configured the remote phone using
STUN and Symetrical RTP. However, the calls dropped every 19-20 seconds. I
read several threads online, but nobody explained the requirements in
details. Everything works fine if you have a public IP address or DMZ on
Asterisk.
Good luck and please let me know if you get it up and running.
Fidel Garcia
System Engineer
sysTeam.
7205 NW 19th Street, Suite 302
Miami, Florida 33126
Email: [EMAIL PROTECTED]
Tel: (305)-477-7303 Fax: (305)-477-0013
http://www.systeamusa.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Friday, June 20, 2008 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice only works from one way.
Yes, both Asterisk and Cisco are behind Nat.
On 6/20/08 3:26 PM, Sam Tam [EMAIL PROTECTED] wrote:
Are you using NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] On Behalf Of Sang-Kil
(Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to call F, and it will ring. Voice from softphone to F carries
over and I can hear it; however, no voice from F to softphone will carry. I
have been experimenting with different codec and other cisco/asterisk config
tips from the web. None had worked so far.
If anyone have experienced such problem and knows how to solve this, I will
be eternally grateful.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very
extensions.conf
[my-phones]
exten = 2000,1,Dial(SIP/2000)
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 2003,1,Dial(SIP/2003)
exten = 6000,1,MeetMe(600,i,54321)
;include = lan-phones
[bogon-calls]
exten = _.,1,Congestion
[pstn-incoming]
include = lan-phones
[local-phones]
include = lan-phones
include = pstn-outbound
[pstn-outbound]
; Calls starting with 9 have the 9 stripped are then routed out to the
PSTN
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten = _9,1,Dial,SIP/[EMAIL PROTECTED] ; IP address of Cisco
gateway
;exten = _9,2,Congestion
exten = _9.,2,Congestion
[lan-phones]
exten = 1001,1,Dial(SIP/1001,20)
exten = 1001,2,Voicemail(u1001)
exten = 1001,3,Answer(SIP/1001)
exten = 1001,102,Voicemail(b1001)
exten = 1001,103,Hangup
Cisco 2611 config
Building configuration...
Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy
