RE: [Asterisk-Users] mpg123 alternative?
Check out the musiconhold.conf.sample in the asterisksource/configs folder. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: 23 February 2006 18:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mpg123 alternative? Ah! Now this is actually something I've not been able to get my head around: > Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which > has its own MP3 player. Can anybody tell me where this built-in MP3 player is in 1.2.x/how do I use it ? I still seem to have the usual two mpg123 processes running with 1.2.4, with whatever music on hold is set in musiconhold.conf I'm sure it is very obvious, but I can't for the life of me figure out what I'm supposed to do to use the built-in MP3 player facilities. I just have the following in my musiconhold.conf: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 random=yes Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hints
Hi Mike
I have build 18 on the IP10's and I have tried call-limit=1 but it still
doesn't work.
Do the extension phones need to have any settings changed to enable this
feature?
Here is my sip.conf:
[11]
callerid="Reception" <11>
username=11
secret=pbx
type=friend
host=dynamic
dtmfmode=rfc2833
mailbox=11
subscribecontext=internal
callgroup=1
pickupgroup=1
notifyringing=yes
qualify=yes
[12]
callerid="Boardroom" <12>
username=12
secret=pbx
type=friend
host=dynamic
dtmfmode=rfc2833
mailbox=12
callgroup=1
pickupgroup=1
call-limit=2
notifyringing=yes
qualify=yes
And a snippet of extensions.conf:
[internal]
exten => 11,hint,SIP/11
exten => 11,1,Macro(dial-extension,11)
exten => 12,hint,SIP/12
exten => 12,1,Macro(dial-extension,12)
exten => _0X.,1,Macro(dial-external,${EXTEN:1}) ;
External calls
How does the hints work? Do you know anything about the flow?
Thanks
Garth
Mike Pollitt wrote:
Hi Garth --
Other users have also reported problems with the status being set by the
SwissVoice phones - oh wait a minute... that was you!
Have you tried setting call-limit=1 in sip.conf? Also check that you're
running the latest firmware on the Swissvoice (I think it's build 18), since
I know they've been tinkering with the presence features recently.
Cheers,
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garth van
Sittert
Sent: Thursday, 23 February 2006 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk hints
I am using Swissvoice IP10S phones.
Garth
Mike Pollitt wrote:
Garth --
What kind of phones are you using?
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garth van
Sittert
Sent: Wednesday, 22 February 2006 7:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk hints
Hi All
Does anyone know how the hints in asterisk works? How does a SIP phone
interact with the hints? I am having a problem with certain phone
models that do not set the hints correctly when I list the hints with a
'show hints'.
Thanks
Garth
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Re: [Asterisk-Users] Codec order sent wrong from Asterisk
hi Palma, as the SJ initiate the call, it will allways go with GSM Codec as the codec should be identical used on both sides. as you do not have G729 on the SJ, it will never use G729. furthermore, i think that if the GSM will not work, then the second option choosed would be PCMA i hope i gave you a way further. Mickey On 2/23/06, Álvaro Palma <[EMAIL PROTECTED]> wrote: I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000.The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMUGXP2000: G729 - GSM - PCMA - PCMU(I have a G729 license, so there's no problem with transcoding G729)In my sip.conf, I've defined the following codec order: disallow=allallow=g729allow=gsmallow=g726allow=alawallow=ulawAnd my peers shows this order correctly:Codecs : 0x11e (gsm|ulaw|alaw|g726|g729)Codec Order : (g729,gsm,g726,alaw,ulaw) Canreinvite is set to NO.But, if I initiate a call from the softphone to GXP-2000, Asteriskalways to the GXP phone GSM as the first codec choice, instead of G729,as I could check with ethereal running in the same server than Asterisk. The SIP INVITE from Asterisk to GXP looks like*Request-Line: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Message HeaderMessage bodySession Description ProtocolSession Description Protocol Version (v): 0Owner/Creator, Session Id (o): root 27682 27682 IN IP4 192.168.1.2Session Name (s): sessionConnection Information (c): IN IP4 192.168.1.2Time Description, active time (t): 0 0Media Description, name and address (m): audio 14224 RTP/AVP 3 18 111 8 0Media Type: audioMedia Port: 14224Media Proto: RTP/AVPMedia Format: GSM 06.10Media Format: ITU-T G.729Media Format: 111 Media Format: ITU-T G.711 PCMAMedia Format: ITU-T G.711 PCMUMedia Attribute (a): rtpmap:3 GSM/8000Media Attribute Fieldname: rtpmapMedia Attribute Value: 3 GSM/8000 Media Attribute (a): rtpmap:18 G729/8000Media Attribute Fieldname: rtpmapMedia Attribute Value: 18 G729/8000Media Attribute (a): fmtp:18 annexb=noMedia Attribute Fieldname: fmtp Media Attribute Value: 18 annexb=noMedia Attribute (a): rtpmap:111 G726-32/8000Media Attribute Fieldname: rtpmapMedia Attribute Value: 111 G726-32/8000Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmapMedia Attribute Value: 8 PCMA/8000Media Attribute (a): rtpmap:0 PCMU/8000Media Attribute Fieldname: rtpmapMedia Attribute Value: 0 PCMU/8000 Media Attribute (a): silenceSupp:off - - - -Media Attribute Fieldname: silenceSuppMedia Attribute Value: off - - - -* So it can be clearly seen how GSM is before G729.Anybody knows if this is an existing bug? Or am I doing something wrong?Thanks a lot for your attention.--Atly.Alvaro Palma___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] IAX2 through Shorewall rpoblem
Hello the list, Be carefull to have this rule available at begining of your rules list, because shorewall use the first one matching and stop to check the following. If you have another with a range including this UDP 4569 DNAT before your new one (as UDP 1024 to 65535 for example), it could shortcut it definitively... Best Regards, Francois BERGERET, France -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rich Adamson Envoyé : jeudi 23 février 2006 12:41 À : Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Objet : Re: [Asterisk-Users] IAX2 through Shorewall rpoblem > I am trying to put a Shorewall firewall in front of my PBX, all the > other port forwards work fine but forwarding port 4569 to the PBX is not > working, it is being logged as rejected even though there is a DNAT rule > in shorewall. > Anyone seen this and have a solution? Are you sure its forwarding "udp" 4569 and not "tcp"? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
On 2/22/06, Clint Sharp <[EMAIL PROTECTED]> wrote: > I had to drop 1.0.1.12 because it has a serious handset volume issue that > seems to cut the handset volume in half. Fix one bug, cause another. True, but the latest (beta, okay, but does that matter?) firmware fixes bot and some other. Please watch the voip-info wiki to check the current status, but it seems to be heading the good way.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which Quad Port FXO is Best?
I'm looking to handle 3 PSTN lines with my Asterisk server. I have a Digium TDM22B and Sipura 3000. The Sipura works great, but the TDM22B seems to have terrible problems with my board---virtually all peripherals need to be disabled in BIOS, and then there is terrible noise, terrible silence and virtually no ability to use DTMF in IVRs. I really wish the TDM22B worked, because I much prefer storing all my configurations on one device, and not needing separate peer accounts for each PSTN line. However, I don't have the skills or spare hardware to debug this quickly, and I'm really wanting to get on with the task of developing some AGI apps. I see several 4 port FXO Analog/SIP gateways on voipsupply.com: [$350] Clipcom 410: http://www.voipsupply.com/product_info.php?products_id=240 [$635] Mediatrix 1204: http://www.voipsupply.com/product_info.php?products_id=171 [$560] Patton 4114: http://www.voipsupply.com/product_info.php?products_id=863 I know it would be cheaper to buy two more Sipuras, but it might be worth the extra $$ to cut down on the power adapters and have a centralized point of administration, especially if it didn't involve dozens of browser mouse clicks to 3 separate HTTP servers. Reliability is the primary criterium, though. Can anybody give any recommendations? And are these digium problems unusual? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp debug or logging
Maybe this is a stupid question but how to you enable debubg or logging on spandsp? I see you can do that for RXFAX but when people tell you to enable debug on spandsp, do they mean this with rxfax or how do you do it with spandsp? I have read that sometimes your faxes come as garbage due to problem with timing, do this affect only E1/T1 or also analog like TDM400P? Thx Poeple! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax receive using TDM400P
Guys. Ive been testing how to receive faxes using TDM400P cards and so far, after playing with gains, echocancell and echotraining on zapata.conf.. Ive ha dno luck, faxes come in as garbage or broken or with blank lines. Anybody has successfully done this? Any tips.. Also I have some ideas: 1. Is it really possible to get fxes on a fax machine using ATAs like the sipura 2002? Even using ulaw and pass-thru, is it possible? 2. Since the faxes is coming from PSTN thru the card, I guess asterisk will always stay in the middle right? No way around this. 3. Im also trying to receive faxes usign a TE110P card with spandsp, unicall and E1 R2MFC, no luck also, some stuff, garbage and broken faxes. Anybody done this sucessfuly? Hope anybody can share their thoughts and insight on this. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] digium TE405P and intel motherboard
The right direction is here: http://www.sangoma.com/datasheets/p_aft-104d-specs?PHPSESSID=82b00b2122ed47a4ac6f4f56487d740f Subject: [Asterisk-Users] digium TE405P and intel motherboardHi,Can please someone help me. I have successfullyinstalled Asteriskathome 2.5 on a server with a IntelServer Board SE7525RP2. May issue is after placing theTE405P in the server, it is not booting anymore. Hasanyone in here have the same experience? Can someoneplease point me to the right direction. Thanks in advance,Leonimar Brings words and photos together (easily) with PhotoMail - it's free and works with Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skype-to-Asterisk(SIP): progress
I want to try this http://www.rsdevs.com/psgw_sip3.shtml but is it worth spending $40? Feedback to the list from any one who tried would be useful. On 2/1/06, John Todd <[EMAIL PROTECTED]> wrote: > > > The developer has indicated that a revised version of the > >> PSGW (http://www.rsdevs.com/) code will be available for sale > >> shortly with the changes. > > > >Has the developer indicated to you whether this would be a free upgrade for > >existing clients or whether additional payments would be expected? > > > >Regards, > > > >Chris > >-- > >C.M. Bagnall, Director, Minotaur I.T. Limited > >This email is made from 100% recycled electrons > > No, there has been no indication of this but it hasn't been > discussed, either. If you are a registered subscriber, perhaps it > may be best to ask. It sounds reasonable to _me_ at least... > > JT > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I force Asterisk t not override my codec order?
Actually no. As far as I understand it, the receiving station gets to dictate the codec used. You call and offer up your list. He selects his preffered from your list and off you go. in your case you will always have gsm from 1>2 becasue 2 has a prefference for GSM. Try it back the other way. You should get an alaw connection Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Álvaro Palma wrote: I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no): softphone 1 <- PCMA -> Asterisk <- GSM -> softphone 2 But, strange for me, Asterisk forces both sides to GSM. I guess that this feature is done to avoid the problem of users setting always the more bandwith consuming codec against its administrator desires. However, is there a way to bypass this feature, so Asterisk set as codec order the same offered by the softphone? Something in sip.conf like: use_client_codec_order = yes (no by default)??? Thanks a lot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicemail problems
"Mustardman", Just call up the voicemail app with the u or b option, as in: Exten => 1,1,Voicemail([EMAIL PROTECTED]|u) Mike > I'm having a similar problem where I keep getting the initial configuration > menu even though I already gone through it and recorded all my greetings. > How do you configure the b or u flag? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] anyway to a2billing without IVR
Hello list, Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk). I want to dial the destination number to the asterisk. for example: user dials, exten =>_011.,1,DeadAGI(a2billing) system will connect the destination and bill them. but right now we need to enter the destination followed by the IVR prompts which i dont want. Thanks in advanved if anybody can help me. best regards shaon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username as extension
On 24/02/2006, at 5:40 AM, Mike Pollitt wrote: You want regexten/regcontext in sip.conf under each peer. http://www.voip-info.org/wiki-Asterisk+sip+regcontext Yes, thats exactly what I want. the scenario is i have two asterisk boxes. box one (10.0.1.1) has a phone that registers as 1000 box two (10.0.2.1) has a phone that registers as 2000 I want to be able to call [EMAIL PROTECTED] and [EMAIL PROTECTED] Now i realise I can add an extension similar to this exten => 1000,1,Dial(Sip/100) But I just want it to somewhat dynamic so if 1000 registers on 10.0.1.1 I can easily send it a call with the Dial(SIP/[EMAIL PROTECTED]) from box two (or another box). regexten is basically what I want but as stated earlier is has some shortcomings in the face that you must restart (not reload) asterisk when the state of a sip peer changes. Regards, Nathan. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GPS-enabled cell phone/PDA
I would like to capture the lat/lon coordinates from a GPS-enabled cell phone or PDA. Is this possible? Must I subscribe to this information from the cellphone network provider, or can I capture it without charge? What devices will broadcast the coordinates? Is there a device that will broadcast its position inband that can be captured by Asterisk? Can an SMS message include coordinates? The subject will willingly carry the device and will be aware that his location is being monitored, so privacy rights are not an issue. The subject will make periodic calls to the Asterisk server in order to record his movements. Does anyone have experience in this area? Thanks, Mike -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context being ignored by inbound sip call
On Feb 23, 2006, at 10.43, btb wrote: Johnathan Corgan wrote: btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = "ipkall incoming" <7508> nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/ from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. thanks jonathan- i originally had this entry as type=user, and switched to type=peer after finding the context was being ignored and reading that type=user may/is be(ing) phased out: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer i've tried type=user again (as well as type=peer), with some additional parameters (mostly guesses, because i don't yet fully understand registration): [7508] ;ipkall type = peer host = dynamic dtmfmode = rfc2833 context = remote callerid = "ipkall incoming" <7508> nat = no insecure = very i gather the ideal method is to know the source ip and source port of the connection from my peer, and include that in the sip config? how can i make asterisk tell me where a connection is coming from? so, in answer to my own question, this ended up being what i needed in sip.conf: [ipkall] type = peer host = voiper.ipkall.com dtmfmode = rfc2833 context = remote callerid = "ipkall incoming" nat = no the key was the host parameter. as soon as i added that, matching occurred and the context was honored. thanks -ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor a call in progress.
> A client call. > A user answer. > Another user, a manager, for instance. Dial a code: > > For instance: > > exten => 1010,1,() #Start to listen the call placed in the channel 1 > exten => 1011,1,() #Start to listen the call placed in the channel 2 > > And so on... What you are looking for is this : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ZapBarge unless you client call isn't coming on a zap channel. In that case, you should look here : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanSpy hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] digium TE405P and intel motherboard
Hi, Can please someone help me. I have successfully installed Asteriskathome 2.5 on a server with a Intel Server Board SE7525RP2. May issue is after placing the TE405P in the server, it is not booting anymore. Has anyone in here have the same experience? Can someone please point me to the right direction. Thanks in advance, Leonimar __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mysql problems
My database machine is broken and I have to use another one.
I made somewhere mistake(s) and get now in the debug file:
[Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query: SELECT * FROM
sip_buddies WHERE name = '886'
[Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query Failed because:
Can't find file: './astconf/sip_buddies.frm' (errno: 13)
[Feb 24 09:05:25] DEBUG[32664]: MySQL RealTime: Retrieve SQL: SELECT *
FROM sip_buddies WHERE name = '494'
[Feb 24 09:05:25] DEBUG[32664]: MySQL RealTime: Everything is fine.
and in CLI:
[Feb 24 09:23:16] WARNING[32664]: res_config_mysql.c:135 realtime_mysql:
MySQL RealTime: Failed to query database. Check debug for more info.
[Feb 24 09:23:16] NOTICE[32664]: chan_sip.c:10935
handle_request_register: Registration from '"886"
' failed for '61.218.43.42' - Username/auth name
mismatch
I have added in the mysql user file (the line what I got from the old
database server with mysqldump):
INSERT INTO user VALUES
('192.168.250.20','','0aaaf','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y','','','','',0,0,0);
192.168.250.20 is asterisk
192.168.250.33 is the old database server
192.168.250.254 is the new database server
I have restarted the database server
I have edited the files:
cdr_mysql.conf
[global]
;hostname=192.168.250.33
hostname=192.168.250.254
res_mysql.conf
[general]
;dbhost = 192.168.250.33
dbhost = 192.168.250.254
dbname = astconf
and restarted asterisk
What do I miss?
bye
Ronald Wiplinger
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[Asterisk-Users] maxmessages and maxgreet per mailbox
From voicemail.conf: ; Maximum number of messages per folder. If not specified, a default value ; (100) is used. Maximum value for this option is . ;maxmsg=100 ; Maximum length of a voicemail message in seconds ;maxmessage=180 ; Maximum length of greetings in seconds ;maxgreet=60 I would like to configure these parameters on a per mailbox basis using Realtime voicemail. I was successful at getting maxmsg to work by adding a field to the realtime voicemail table when using Asterisk 1.2.4: CREATE TABLE `voicemail` ( `uniqueid` bigint NOT NULL auto_increment, `customer_id` bigint NOT NULL default '0', `context` varchar(50) NOT NULL default '', `mailbox` bigint NOT NULL default '0', `password` varchar(5) NOT NULL default '0', `fullname` varchar(150) NOT NULL default '', `email` varchar(50) NOT NULL default '', `pager` varchar(50) NOT NULL default '', `tz` varchar(10) NOT NULL default 'central', `attach` varchar(4) NOT NULL default 'yes', `saycid` varchar(4) NOT NULL default 'yes', `dialout` varchar(10) NOT NULL default '', `callback` varchar(10) NOT NULL default '', `review` varchar(4) NOT NULL default 'no', `operator` varchar(4) NOT NULL default 'no', `envelope` varchar(4) NOT NULL default 'no', `sayduration` varchar(4) NOT NULL default 'no', `saydurationm` tinyint(4) NOT NULL default '1', `sendvoicemail` varchar(4) NOT NULL default 'no', `delete` varchar(4) NOT NULL default 'no', `nextaftercmd` varchar(4) NOT NULL default 'yes', `forcename` varchar(4) NOT NULL default 'no', `forcegreetings` varchar(4) NOT NULL default 'no', `hidefromdir` varchar(4) NOT NULL default 'yes', `stamp` timestamp(14) NOT NULL, `maxmsg` int NOT NULL default '60', PRIMARY KEY (`uniqueid`), KEY `mailbox_context` (`mailbox`,`context`) ) TYPE=MyISAM ; Is it possible to do the same for maxgreet and maxmessage? And if not in Asterisk 1.2.4, is there a patch available? Any help or suggestions are greatly appreciated. Thanks, Anish Basu ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming/Outgoing call question
For handling outbound calls, your easiest approach is to have each business'
phones default to different contexts, so you might have the dialplan
arranged as follows:
[in-pstn]
exten => number,1,dosomething
exten => number,1,dosomething
exten => number,1,dosomething
etc.
[business1]
; internal calls within business1
exten => _2XX,1,Dial(SIP/business2-${EXTEN})
[business2]
; internal calls within business2
exten => _2XX,1,Dial(SIP/business2-${EXTEN})
[business3]
; internal calls within business2
exten => _2XX,1,Dial(SIP/business3-${EXTEN})
Then of course in the appropriate sip/zap/iax.conf files where you've got
the various business' phones defined, set context=business1/2/3 as required.
To handle incoming calls from PSTN numbers I'd define 3 queues containing
the users from each business, then set each of the incoming numbers in
[in-pstn] to go to the appropriate queue. You'd probably also want to
include in-pstn in each business' context for outbound calls so that calls
made between them to each other don't go all the way out onto the PSTN and
back again.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
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Re: [Asterisk-Users] Incoming/Outgoing call question
On Thu, 2006-02-23 at 17:53 -0500, Kevin Smith wrote:
> Hey everyone,
>
> I have a more of an opinion question then a technical question. The
> asterisk server I am setting up is going to host 3 different businesses.
> Each business is in the same building, and on the same network. My
> question is regarding calls coming in and going out. We are a small ISP
> and have a lot of numbers that are forwarded to our phone system. The
> other companies have about 3 to 5 numbers going into their offices. My
> question is if there is a good way to test for which number and where to
> send it to.
>
> Right now my though process was something like this (keep in mind I
> haven't wrote it):
>
> [default]
> include => Our-Numbers
> include => Business1
> include => Business2
>
> [Out-Numbers]
> exten => s,1,gotoif,$[${EXTEN}=Number1 |
> ${EXTEN}=Number2..${EXTEN}=NumberN]?Match:1|:
>
> Is that the best way to test for the number that is being dialed? Or can
> you recommend a better way. If anyone has done something similar could
> you share how you did this type of a setup? I know I could manually put
> in each one, but I think there probably is a better way. If I have to go
> that route, then I probably will write a script to generate the file.
>
> Thanks,
> Kevin
>
For incoming calls I'd do something like (simplified):
[incoming-calls]
exten => BUSINESS1,1,Dial(SIP/business1)
exten => BUSINESS2,1,Dial(SIP/business2)
...
assuming you got channels that give you that information, e.g. ISDN or
so. Analogue pots won't provide the information which number was
dialled.
Asterisk will match the number that was dialled in the dialplan as the
extension. You might want to put a pattern in in case the number dialled
isn't specifically listed. (e.g. your numbers?)
Conrad
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RE: [Asterisk-Users] How to query a table from the keypad?
Hi Richard – What you want is AGI: http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI You could write a perl script to read the PO number from stdin and spit back the balance (or whatever). To read the PO number from the user, use the Read() dialplan application. To play back the balance, you could use SayDigits() (but there’s probably a more elegant solution specifically for speaking amounts of money). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Reina Sent: Friday, 24 February 2006 9:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to query a table from the keypad? I am trying to give users the option to query our accts. payable database by supplying their PO number. I able to write queries via perl->DBI->mysql but have no idea how to get * to do it from the IVR. Is this possible? Can anyone point me in the right direction for help or examples? Thanks, Richard What are the most popular cars? Find out at Yahoo! Autos ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
On Thu, 2006-02-23 at 15:48 -0700, Colin Anderson wrote: > >The cost saving of being able to pin-point a cabling/NIC/bandwidth > >problem down to the port on the switch easily and quickly is wonderful > > We also use 3com NJ-200's which is a 4 port switch in a wall plate that has > SNMP and other goodies. I can troubleshoot down to the wall plate, anywhere > in the world. Last year I was on holidays in Vancouver (1000K away from the > office) and I got the call that an exec couldn't plug his laptop into the > wall, no signal, and he was pissed. I whip out my laptop, walk across the > street to Starbucks, got a wifi signal, VPN in, I check it out - nope, it's > your stupid laptop, PHB-boy. Turns out he disabled the onboard NIC. That > single incident, to me justifies the whole expense of a good infrastructure > (and to the PHB too - he was spooked that I could do that) > ___ I used to have VNC and TopgunSSH on my Palm and an Infred connection to my mobile and from there on to the internet ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to create channel of type 'SIP'
Hi, i have a working asterisk version CVS-v1-0-05/13/05-15:06:32, i was installed using amportal , i want to migrate to another server, this time i dont wat to use amportal and edited "by hand" everyfile, i can make outboundcalls without problems, but i cant receive anything, either from between the sip phones or the external peers, i copied the sip.conf from the old server, this is the relevant port of the external peer, a cisco as5400: ### sip.conf ### [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=g729 allow=g723.1 allow=alaw allow=ulaw context = bogon-calls ; Send unknown SIP callers to this context callerid = Unknown language=es register => @prepago-in [prepago-in] type=friend host=aaa.bbb.ccc.ddd context = from-external dtmfmode=rfc2833 insecure=very ; required for incoming FWD calls [prepago-out] type=peer ; we only want to call out, not be called host=aaa.bbb.ccc.ddd dtmfmode=rfc2833 [22662124] callerid="22662124" <22662124> context=from-internal host=dynamic secret=22662124 type=friend username=22662124 this is the error log Destroying call '[EMAIL PROTECTED]' Feb 23 19:16:52 NOTICE[1023]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Transmitting (no NAT) to aaa.bbb.ccc.ddd:5060: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP aaa.bbb.ccc.ddd:5060;received=aaa.bbb.ccc.ddd From: ;tag=1FA0C538-A3B To: ;tag=as3aaf6cd5 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: No route to destination what can we wrong? --- Miguel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Voicemail problems
I'm having a similar problem where I keep getting the initial configuration menu even though I already gone through it and recorded all my greetings. How do you configure the b or u flag? > -Original Message- > From: Michaël Gaudette [mailto:[EMAIL PROTECTED] > Sent: Thursday, February 23, 2006 1:30 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Re: Voicemail problems > > Thanks Rich and CF for responding to my query. > > Turns out that I wasn't using the b or u flag to define > whether the unavailable message or busy message should be > played. By doing that, I fixed my issue. Thanks Rich. > > I really do think that Asterisk should have some sort of > logic that chooses which message should be played (when one > has been recorded). Is there a reason that escapes me that > Asterisk chooses the generic message when it isn't told which > message to pick? > > > Mike > > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
Are you sure those switch figures are right? 16ms delay in the switch path sounds a bit long. Cisco's mid-range switches like the 2950 have switching times measured in micro seconds. Then again a 2626 procurve is only around $700. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Conrad Wood Sent: Friday, 24 February 2006 7:50 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] What business IP phone to use > Simple formula: > > 1. Total Revenue > 2. % of revenue derived from phone usage > 3. =Cost of downtime by using SoHo or consumer gear. > > It's not a question of if a SoHo or low cost device will screw up, it is a > question of when. This is 23 years of experience talking. > > Where I work, the value of #3 above is $16 Cdn a *second*. We are below 500 > employees, so we fall into the SMB segment. Sometimes I'm appalled by > statements that a $700 switch or a $400 phone isn't worth it. Huh?? Maybe in Absolutely right! for something as critical as switches & cabling I always recommend to spend real money. Don't ever try to save money any equipment that is required to operate the business. (Had very good experience with HP procurves over the last 10 years or so). There is no point buying netgear or other low-cost switches for a business ever. The cost saving of being able to pin-point a cabling/NIC/bandwidth problem down to the port on the switch easily and quickly is wonderful. Combined with SNMP and all the other goodies good switches come with, our clients save a lot of money by paying me less for my time ( d'oh ;-) ). The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! I see no reason however to spend $400 on a single phone though, because if a single phone breaks, it's not going to bring your business to a standstill, is it? (I guess unless you only have one in the first place ;-) ) conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
> >> After 2 weeks of messing around with every conceivable IAX2 and > >> jitterbuffer configuration, I switched to SIP yesterday. Complaints > >> went from 10-20 per day to ZERO. Literally overnight. > > > > I wonder if this is an ILBC frame size issue of some sort? Seems odd. > > I've got to add my name to the list here. We're just using GSM over our IAX > links, and our jitterbuffer values look like this: > > maxjitterbuffer=1000 > resyncthreshold=1000 > maxjitterinterps=10 > > For the most part the new jitterbuffer actually yields much better quality > than the old jitterbuffer, but when the resyncs happen, it's like the call > has a lot of trouble getting get back on track. It flounders for quite a > while, with badly broken audio, sometimes up to 20 seconds before coming > back. I've tried hanging up as soon as event starts happening and then > immediately calling the same number, and the channel comes back with crystal > clarity. So it seems to me like there is something askew with the resync. If memory serves correctly, I believe I remember Mark applying a fix to the iax jitterbuffer and that fix had something to do with a counter rollover or something like that. That fix happened in the last week or so. I'm not sure if that would have been included in v1.2.4 or not, but might be worth a little research. I also opened a bug a month or two ago involving ilbc and iax, and someone else confirmed it was a bug. Don't have the bug number handy, but the problem related to a combination of iax trunking, jitterbuffer and ilbc. Disabling one of those consistently bypassed the problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: VoIP over bonded link
Colin Anderson a écrit : It's stupid. Don't ever connect 2 different building with copper. Just wait until you get some kind of lightening hit or electrical fault, but make sure you are no where near it. Use fibre. Thanks for the reply. Unfortunately, the conduit for the provisioning of the new building is unsuitable for fibre (too many sharp bends) and we can't core out the concrete and put in a new conduit because of obstacles in the way that make laying new conduit impractical, so we are stuck with (existing) copper. We already have copper-to-copper connections of different types (electrical, security etc) between the buildings so a lightning strike is going to hose us no matter what. That aside, does anyone have opinions on my original question as to the suitability of bonded links for VoIP? You might have a little bit of jitter, but that's what jitter buffer is all about. IMHO it would be fine for VoIP but as it has been pointed on the list it would be wise to prioritize correctly both ends of your aggregated link. PS: What about Wifi for your link? With a couple of well placed high-gain outdoor antennas you could cover the distance and have similar throughput... It could be significantly cheaper too! Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup call on Hold
On Thu, 2006-02-23 at 11:08 -0500, Sean Cook wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Is it possible to pickup a call that is on hold on another extension? > Does anyone have any magic they can share on this topic? > > I am struggling to teach call parking at a local shop where we installed > *. It would simplify my life so much if they could just put the call on > hold and pick it up on another line. Have you tried the parking application? Depending on what phone you use you might be able to reprogram the "hold" button. http://www.voip-info.org/wiki-Asterisk+call+parking Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicemail problems
Michaël Gaudette wrote: > I really do think that Asterisk should have some sort of logic that chooses > which message should be played (when one has been recorded). Is there a > reason that escapes me that Asterisk chooses the generic message when it > isn't told which message to pick? It plays the person's name when no greeting is selected, and if you set up the options to make them record their name during mailbox setup. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: VoIP over bonded link
Colin Anderson wrote: > That aside, does anyone have opinions on my original question as to the > suitability of bonded links for VoIP? It's not an issue. The bonded link acts just like any other link. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended rack-mountable server anyone?
Yusuf, Could you find out the brand/make of the servers you used? That would be very helpful for me. Thank you! Stagg, Those servers sound like they should be avoided at all costs... Thank you for the heads up ;). On 2/23/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Hi, > > we have used about 6 Intel rack mounted servers, with dual Xeon > processors, (I have forgotten the exact make.) we used them with Digium > quad span pri cards, and some with Sangoma cards, works well . These > servers are still pretty solid. they come with two onboard NICs, and with > the Digium cards, one of the NICS had to be disabled cause of interrupt > clashes. Other wise they have worked well. > > > I forgot about one other issue we had with the 2850. The integrated > > NICs caused interrupt issues with the TE411P. We had to disable the > > integrated NICs, and installed dual port gigabit intel NIC. > > > > Stagg Shelton > > www.oneringnetworks.com > > > > [EMAIL PROTECTED] wrote: > > > >>Alexander, Perhaps I'm wrong, but I have a server here next to my desk > >>(IBM e325) and I tried to fit a normal pci card into it. The slots are > >>completely different and the card would not fit.. this was just a pci > >>dvi video card. The server specs say that it is using PCI-X technology > >>for the slots so this leads me to believe that they are not compatible > >>as one would think. > >> > >>Cory Andrew, I will look into the supermicro servers again, I'm not > >>keen on the handles up front on them though, that makes for awkward > >>handling (imo). > >> > >>Wow Stagg, Thank you for that first hand knowledge. These are things > >>you just can't learn until you buy a product and experience it first > >>hand. I'm not so sure that we want to Frankenstein our own cable for > >>this configuration though! (yikes!) > >> > >>Hopefully some other people will pipe up too with some more server > >> suggestions! > >> > >>On 2/22/06, Stagg Shelton <[EMAIL PROTECTED]> wrote: > >> > >> > >>>We just installed asterisk for a customer using a Dell 2850. It has 3 > >>>pci slots. My customers configuration contained a TE411p Quad Span PRI, > >>>and a TDM400P with 4 FXS Modules. The only problem that we had with the > >>>2850 was getting power to the TDM400P. We located a power connector on > >>>the backplane that supplied the required 12v. I think it was originally > >>>intended to power a tape backup drive. We ultimately sacraficed a power > >>>supply to get at it's 12V P4 connector. We then used a voltmeter to put > >>>together a pinout for the dell power port, and frankensteined together a > >>>cable that could be used to power the TDM400P.Aside from the power > >>>issue, the platform seems rock solid. > >>> > >>>Stagg Shelton > >>>www.oneringnetworks.com > >>> > >>> > >>>[EMAIL PROTECTED] wrote: > >>> > >>> > >>> > Hey everyone, > > I've been doing a lot of research into a decent server for Asterisk > but I seem to be running and circles and now I am turning to you. The > issue I have is it needs to be rack mountable (so a Dell SC430 isn't > going to work) and preferably have 3 pci ports. The problem that I > seem to be running into is that when I look at servers from Dell or > IBM or the like they only seem to support PCI-X which (from what I > understand) does not support the Digium cards that we already have and > that they still make. So if anyone has a suggestion or has a server > they rather prefer for it's reliability, expandability, etc, please > recommend it! > > Thank you in advance, > Mitchel > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > >>>___ > >>>--Bandwidth and Colocation provided by Easynews.com -- > >>> > >>>Asterisk-Users mailing list > >>>To UNSUBSCRIBE or update options visit: > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >>> > >>> > >>___ > >>--Bandwidth and Colocation provided by Easynews.com -- > >> > >>Asterisk-Users mailing list > >>To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >> > >> > > > > -- > > This message has been scanned for viruses and > > dangerous content by MailScanner, and is > > believed to be clean. > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > > ___
Re: [Asterisk-Users] OT: VoIP over bonded link
Colin Anderson wrote: I have to provision several dozen * users to a seperate building on our campus in the same subnet. Ordinarily, I'd just run a gigabit cat6 cable to another switch if it doesn't violate the 100 metre rule, but this building is several hundred metres away from my backbone. My only option for cabling to the remote building is copper. My plan is to provision them with a Linux bridge with 4 NIC's: 1 gigabit to the backbone, and three bonded together as a single interface (90 mbit aggregate), then plugged into this dealie: http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946 At the remote building, the reverse: another Linux box with 4 NIC's that de-aggregates the link to a gigabit connection on a switch, and then to the wall plates. I'm pretty sure this will work for data no problem, but I'm a little concerned about latency on a timing-sensitive applicaiton like VoIP. Anyone have experience with VoIP over bonded link? Is there a gotcha? Is this a stupid idea? On my whiteboard it looks fine! If you have line of site, or even close, you can consider running VoIP over wireless bridges. We've run VoIP and network traffic over the Cisco 1300 and 1400 series bridges with no problems. They will support voice VLANs and qos. Thanks, Nick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analyzer for Milliwatt
Hi, app_milliwatt is a nice tool for a quick check of the line quality. Anyway, hearing to that tone for more than a minute is painful. Does anyone know the "opposite" application, i.e. an application, that "hears" and listens for a 1000 Hz tone and displays the quality in any unit? If not, I'll think about developing one. Regards, Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
We switched to that after mpg123 wouldn't compile on our newer 64bit machines You might have successfully compiled with "make linux-devel" perhaps. I did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming/Outgoing call question
Hey everyone,
I have a more of an opinion question then a technical question. The
asterisk server I am setting up is going to host 3 different businesses.
Each business is in the same building, and on the same network. My
question is regarding calls coming in and going out. We are a small ISP
and have a lot of numbers that are forwarded to our phone system. The
other companies have about 3 to 5 numbers going into their offices. My
question is if there is a good way to test for which number and where to
send it to.
Right now my though process was something like this (keep in mind I
haven't wrote it):
[default]
include => Our-Numbers
include => Business1
include => Business2
[Out-Numbers]
exten => s,1,gotoif,$[${EXTEN}=Number1 |
${EXTEN}=Number2..${EXTEN}=NumberN]?Match:1|:
Is that the best way to test for the number that is being dialed? Or can
you recommend a better way. If anyone has done something similar could
you share how you did this type of a setup? I know I could manually put
in each one, but I think there probably is a better way. If I have to go
that route, then I probably will write a script to generate the file.
Thanks,
Kevin
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Re: [Asterisk-Users] mpg123 alternative?
Rich Adamson wrote: > Is there any particular reason for the native file format stuff to be in > asterisk-addons as opposed to that code being merged into trunk? It isn't. You are mis-interpreting the information in this thread (it's been unclearly stated anyway). The only portion that is in asterisk-addons is format_mp3, which allows Asterisk to natively open MP3 files. However, that is of little use, when you can use sox to convert those files into slinear/ulaw/alaw/gsm/etc. so that no transcoding is needed when the audio is played to a caller. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
Thanks for the info Matt. I had the zombie issue too. Based on my observations I decided those processes were due to safe_asterisk not killing rawplayer (it does mpg123 processes). So I modified safe_asterisk to do that, and zombie processes were gone. But you may be right, native players may be better. Kevin should know better anyway. (However, on my slooow test machine, native mp3 player speeds up and down sometimes, so I need every processor cycle. Not a recommended configuration on production systems of course.) - Original Message - From: "Matt Roth" <[EMAIL PROTECTED]> Soner Tari wrote: I vote for the raw file format, due to the reasons listed here: http://www.orderlyq.com/asteriskqueues.html Of course you need to convert all mp3 moh files to raw format manually, but it's easy as described there. We were using the rawplayer method on our server, but it ended up spawning hundreds of zombie processes. I talked to Kevin Fleming about it, and he recommended switching to native MOH. Scalability is a big concern of mine, so I asked him about the impact it would have on CPU, memory, and disk utilization. His response was, "not much, more memory usage though." I could spare the memory, so I gave it a shot. So far, I'm very impressed. It eliminated the zombie processes completely and the quality of the playback is *much* better. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What business IP phone to use
>The cost saving of being able to pin-point a cabling/NIC/bandwidth >problem down to the port on the switch easily and quickly is wonderful We also use 3com NJ-200's which is a 4 port switch in a wall plate that has SNMP and other goodies. I can troubleshoot down to the wall plate, anywhere in the world. Last year I was on holidays in Vancouver (1000K away from the office) and I got the call that an exec couldn't plug his laptop into the wall, no signal, and he was pissed. I whip out my laptop, walk across the street to Starbucks, got a wifi signal, VPN in, I check it out - nope, it's your stupid laptop, PHB-boy. Turns out he disabled the onboard NIC. That single incident, to me justifies the whole expense of a good infrastructure (and to the PHB too - he was spooked that I could do that) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to reset Digium card while asterisk is running?
Hi, I currently have a yellow/red alarm on one span of a Digium card. It is not the first time, this already happened some months ago, and I expect to clear the alarm when rmmoding and insmoding the zaptel and wct4xxp modules. Unfortunately I can't rmmod while asterisk is running and I can't stop asterisk, because of lot of traffic on other channels. Ok, I could wait until 2 o'clock in the night, in order to shut down asterisk on low traffic. I wonder, if there is another way to reset a Digium card without the need to shutdown asterisk? Thanks for any hints! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec order sent wrong from Asterisk
Álvaro Palma wrote: > But, if I initiate a call from the softphone to GXP-2000, Asterisk > always to the GXP phone GSM as the first codec choice, instead of G729, > as I could check with ethereal running in the same server than Asterisk. > The SIP INVITE from Asterisk to GXP looks like This is intentional, and well documented on the wiki. Asterisk attempts to use the same codec for the outbound call as the inbound call (if it is allowed), to reduce the need for transcoding. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?
Benchev is believed to have said: >Hi, >Do you have any success receiving the caller id with your TDM400 FXO? >I have the same problem when I connect the GSM gateway to a SPA3000 FXO line >and thought this a Sipura's problem. On a phone connected to the GSM gateway >I can see the callerid, but not on the Sipura's PSTN line ... >Thanks, >benchev > Hello benchev, this is no more and no less the same problem as I do have. It appears it's then not really the TDM400 FXO module. I have another option to test: I do have a similar ATA like the Sipura, but made by Grandstream. It's here at home; I will take it to the office tomorrow and see if it can read the caller id from the GSM gateway. Even my gsm unit does indeed pass the callerID when I connect it to a cheap, dead simple analog phone! BTW: Do you have a manual for the gateway? Best regards, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: VoIP over bonded link
>Some thoughts that you might want to consider... >The vdsl box runs at speeds "up to 15 meg". That translates into the longer >the copper loop, the slower the speed. You'll probably want to accurately >measure the copper loop length and translate that into some 'expected' speed. >Probably won't be 15 meg, and whatever the documentation suggests, it >will likely be a fair amount slower then that. >Does the vdsl truly operate in a full duplex mode with equal bandwidth >in either direction? >We've worked with many corporations and institutions in over 40 states >doing network performance assessments, and seldom (if ever) do bonded >interfaces actually work the way that you might think they work. I've not >spent any time with the linux bonding that you're considering, but you >might want to better understand exactly how that works. E.g., some bonding >actually functions at 'per packet' level, which implies the maximum speed >of any single packet is the speed of one vdsl circuit. >If one of the bonded circuits has errors, what impact does it have on the >other three error-free circuits. (Its not uncommon for one interface to >have very significant impact on all other interfaces.) >If all of the above can be answered with positive thoughts, you'll still >want to consider some form of QoS on those links to ensure the voip >packets are not held in a queue. Good suggestions, thanks. I'm going to build a prototype and do some analysis. If it's good, I'll post to the list. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to query a table from the keypad?
I am trying to give users the option to query our accts. payable database by supplying their PO number. I able to write queries via perl->DBI->mysql but have no idea how to get * to do it from the IVR. Is this possible? Can anyone point me in the right direction for help or examples?Thanks,Richard What are the most popular cars? Find out at Yahoo! Autos ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Choice One PRI?
Hi, all. I've got a T1 through Choice One Communications (www.choiceonecom.com), a provider in the US northeast. I recently tried to switch to ISDN on it -- and failed miserably. I've posted my config files, and nobody's seen anything obviously wrong. Has anyone else used their ISDN T1's? If so, would you be kind enough to send me your zapata.conf and zaptel.conf files? Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems
On Thu, 2006-02-23 at 14:36 -0600, Aaron Daniel wrote: > We use mainly 7940's in our environment, I currently have about 60 of > them on my system, and the 7.5 firmware really screwed up our phone > network. Not in the same way yours has, but there seem to be a lot of > glitches, even jumping from 7.4 to 7.5. Have you tried the 7.4 firmware > to see if that does you any good? > For what its worth, 7.4 seems to work great in my setup, I stayed away from 7.5, luckily I read about the glitching before upgrading. -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: VoIP over bonded link
Take a look at the Net2Net product which was purchased by Paradyne and then by Zhone (www.zhone.com). They make a unit that will bridge Ethernet over SDSL lines, 24 pairs will get you 50mbps through the link. It looks just like ethernet and VoIP will work fine over it. You can also check our RAD (www.rad.com) they have a bunch of long range ethernet extenders to run on existing copper. On Feb 23, 2006, at 3:34 PM, Colin Anderson wrote: It's stupid. Don't ever connect 2 different building with copper. Just wait until you get some kind of lightening hit or electrical fault, but make sure you are no where near it. Use fibre. Thanks for the reply. Unfortunately, the conduit for the provisioning of the new building is unsuitable for fibre (too many sharp bends) and we can't core out the concrete and put in a new conduit because of obstacles in the way that make laying new conduit impractical, so we are stuck with (existing) copper. We already have copper-to-copper connections of different types (electrical, security etc) between the buildings so a lightning strike is going to hose us no matter what. That aside, does anyone have opinions on my original question as to the suitability of bonded links for VoIP? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys WIP300 WiFi Phone
Phil - I found out about the WIP300 not including the documentation CD-Rom, I apologize for that. Linksys is correcting the issue. I just added a download links on the product page for the Manual and the User/Config guide in pdf format, you can find them, below the product image - here http://www.voipsupply.com/product_info.php?products_id=1525 Sorry for the inconvenience. If I see a firmware update that addresses the charging issues you mentioned I will alert you. Thanks, Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: "Philip Edelbrock" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, February 23, 2006 3:08 PM Subject: [Asterisk-Users] Linksys WIP300 WiFi Phone Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit didn't come with a CD-ROM or registration card, eventhough they are listed as being in the Package Contents. - This phone uses a USB port to charge, do firmware updates, and perhaps other things. Sadly... it DOES NOT COME WITH A USB CORD! You'd think for $250 that you'd get a cord included... oh well. It does come with a charger with a usb end so you can charge the phone from an AC outlet, though. - The battery charging animation runs backwards, animating like the battery's charge is flowing out rather than in. A little amusing. No charge status while it is charging, which I don't like. It would be nice to see that it's, say, 75% charged for example. Does anyone else have one of these phones yet? Any gotcha's as far as using it with *? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to install Zaptel?
I greatly appreciate the help at last Monday's installfest and especially am grateful to Chris and Lenny. Now I am reconfiguring the installfest computer to add Zaptel using the instructions at http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation There is now a Digium 4 port card installed. In regards to Zaptel there are some questions on zconfig.h. Should I: #define BOOST_RINGER for more ring voltage? I will have about 12 pots phones #define CONFIG_CALC_XLAW for a small number of channels? I have two incoming dids, two incoming POTS lines and, for now, two outgoing sip channels. Is that a small number? #define CONFIG_ZAP_UDEV for UDEV support? I have kernel 2.6.13 in Slackware 10.2 which uses UDEV - I _think_. Any other things that might need changing in zconfig.h? I'm also looking at the Digium Install Guide (one page pdf file) which says "you will need to have the following to compile Zaptel and Asterisk code" I don't know if the right stuff is already installed or not. Full linux kernel source code (got that of course) zlib devel Is this separate from the zlib files in the kernel source? Should I look for the zlib devel source? OpenSSL (and headers) Do they mean the source or just the executable and headers? I have a bunch of .h header files in /usr/include/openssl but no .c files. Bison 1.875 I appear to have Bison 1.35 but have located Bison 1.875. Too old? Should I compile the new one? My understanding is that full the compile sequence order is: 1. compile unix-ODBC with ./configure --enable-gui=no 2. configure zconfig.h and compile Zaptel 3. compile Pri if you have it (I don't) 4. compile Asterisk: make, make install, make samples, make progdocs Shouldn't make install be _after_ make samples & make progdocs? 5. Asterisk-sounds: make install only What about #4 - doesn't make install come after make samples/make progdocs? Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Slackware Linux ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] auto provision of IP501 polycom
In dhcpd.conf: option tftp-server-name "x.x.x.x" Yes, I know it says tftp but actually this is the entry used for ftp as well. Also, for reasons known only to your chosen deity, Polycom have chosen to use a mixed-case username for the default ftp user. Not all FTP servers will accommodate this. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Friday, 24 February 2006 12:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] auto provision of IP501 polycom Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in the way of dhcp options or dns entries to get the polycom to discover the ftp boot server? What about changing default passwords via ftp? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
Hi - >> After 2 weeks of messing around with every conceivable IAX2 and >> jitterbuffer configuration, I switched to SIP yesterday. Complaints >> went from 10-20 per day to ZERO. Literally overnight. > > I wonder if this is an ILBC frame size issue of some sort? Seems odd. I've got to add my name to the list here. We're just using GSM over our IAX links, and our jitterbuffer values look like this: maxjitterbuffer=1000 resyncthreshold=1000 maxjitterinterps=10 For the most part the new jitterbuffer actually yields much better quality than the old jitterbuffer, but when the resyncs happen, it's like the call has a lot of trouble getting get back on track. It flounders for quite a while, with badly broken audio, sometimes up to 20 seconds before coming back. I've tried hanging up as soon as event starts happening and then immediately calling the same number, and the channel comes back with crystal clarity. So it seems to me like there is something askew with the resync. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 601 Buddy Watch problems
Polycom only supports Asterisk Business Edition. Does ABE even support hints/buddies? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, February 23, 2006 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP 601 Buddy Watch problems On Thu, 23 Feb 2006, Nathan Bowyer wrote: >> It may have something to do with the watch limit on the Polycom firmware. > I have one phone that does this as well, nearly all the time. I believe its > the only one that exceeds the 6-7 buddy watch limit as well. Now that polycom is "committed to working with asterisk", is polycom going to fix the 7 buddy watch limit? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] username as extension
You want regexten/regcontext in sip.conf under each peer. http://www.voip-info.org/wiki-Asterisk+sip+regcontext -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Alberti Sent: Thursday, 23 February 2006 7:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] username as extension Is there a way to have extensions automatically created for registered sip users ? I did some investigation and found some hope in chan_sip with relation to the somewhat undocumented usereqphone option but i may be totally off track. All i want to be able to do is send a call to [EMAIL PROTECTED] where the number is the username configured on the phone that has registered with asterisk on ip_address. From what I understand this should be pretty standard sip functionality no ? Regards, Nathan. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk hints
Hi Garth -- Other users have also reported problems with the status being set by the SwissVoice phones - oh wait a minute... that was you! Have you tried setting call-limit=1 in sip.conf? Also check that you're running the latest firmware on the Swissvoice (I think it's build 18), since I know they've been tinkering with the presence features recently. Cheers, Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth van Sittert Sent: Thursday, 23 February 2006 6:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk hints I am using Swissvoice IP10S phones. Garth Mike Pollitt wrote: > Garth -- > > What kind of phones are you using? > > Mike. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Garth van > Sittert > Sent: Wednesday, 22 February 2006 7:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Asterisk hints > > Hi All > > Does anyone know how the hints in asterisk works? How does a SIP phone > interact with the hints? I am having a problem with certain phone > models that do not set the hints correctly when I list the hints with a > 'show hints'. > > Thanks > Garth > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mpg123 alternative?
>From my understanding of your question, you are looking for the format_mp3 module that is in the Asterisk Add-ons tarball. This is classed as the "native" way of playing mp3 AFAIK Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: Friday, 24 February 2006 5:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mpg123 alternative? Ah! Now this is actually something I've not been able to get my head around: > Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which > has its own MP3 player. Can anybody tell me where this built-in MP3 player is in 1.2.x/how do I use it ? I still seem to have the usual two mpg123 processes running with 1.2.4, with whatever music on hold is set in musiconhold.conf I'm sure it is very obvious, but I can't for the life of me figure out what I'm supposed to do to use the built-in MP3 player facilities. I just have the following in my musiconhold.conf: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 random=yes Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I force Asterisk t not override my codec order?
I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no): softphone 1 <- PCMA -> Asterisk <- GSM -> softphone 2 But, strange for me, Asterisk forces both sides to GSM. I guess that this feature is done to avoid the problem of users setting always the more bandwith consuming codec against its administrator desires. However, is there a way to bypass this feature, so Asterisk set as codec order the same offered by the softphone? Something in sip.conf like: use_client_codec_order = yes (no by default)??? Thanks a lot. -- Atly. Alvaro Palma ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
Matt Roth wrote: We were using the rawplayer method on our server, but it ended up spawning hundreds of zombie processes. I talked to Kevin Fleming about it, and he recommended switching to native MOH. Scalability is a big concern of mine, so I asked him about the I've had issues with Native MOH when using IAX trunking and Placing a caller into a parking slot. Sound is awful. So, for parking I use mpg123 and everything else I'm using Native MOH. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicemail problems
Thanks Rich and CF for responding to my query. Turns out that I wasn't using the b or u flag to define whether the unavailable message or busy message should be played. By doing that, I fixed my issue. Thanks Rich. I really do think that Asterisk should have some sort of logic that chooses which message should be played (when one has been recorded). Is there a reason that escapes me that Asterisk chooses the generic message when it isn't told which message to pick? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK X100P installation help
Hi Paul - We gave up on analogue a long time ago in favour of ISDN. I have 3 ISDN cards in my Asterisk box. Billion ISDN BRI Cards cost me approx £15 each from komplett.co.uk and are perfect. You need to use the bri-stuffed version of Asterisk. If you still have the ISDNline I would recommend you give it another shot. You get none of the echo, caller ID and hangup detection problems with ISDN. It Just Works. (TM) Rgds Tim Robinson Basingstoke, UK. Paul J. Smith wrote: Hi, I've got the latest [EMAIL PROTECTED] setup running with SIP. I've been trying to tie it in with the PSTN off and on for a while with no success. I gave up on ISDN and purchased 2 x100p cards from x100p.com. I've got the card installed, the machine can see it. The problems I have at the moment are First time I start up the drivers and asterisk, I can call the pstn number. The call is detected, the SIP phone rings, you can pick up the call, but there is nothing but a faint noise. After that, if you try to call the number again, it's engaged, as if the x100 P has not dropped the line, though watching the asterisk log in real time it says it has. I've looked all over the web for some detailed instructions, but most just mirror what I've already done. Is it because I'm in the UK? I could not find anyone using these in the UK. Maybe they don't like UK phone sockets? My zaptel.conf is # Span 1: WCFXO/0 "Wildcard X100P Board 1" fxsks=1 # Global data loadzone= uk defaultzone = uk Can anyone help me to get this working? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
I'd suggest using the format_mp3 program that's included in asterisk-addons. We switched to that after mpg123 wouldn't compile on our newer 64bit machines, and it works like a charm, and you don't have to change anything. Aaron Rich Adamson wrote: Been using mpg123 for moh for the last two years or so. However, when I have * config errors, often times get a endless stream of console messages and need to kill the two mpg123 processes. Is there an alternative to mpg123 that eliminates that issue? I see references in musiconhold.conf relative to madplay, native file format, asterisk-addons, etc. Not sure why the asterisk-addon approach hasn't been moved into trunk, or if madplay is a better choice on this fc3 trunk box. Any suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems
C F wrote: I recently updated my phones Cisco 7960 phones (3 of them) in a high volume call place, where the Secretaries use the 7960 phones to answer inbound calls, as many as 15 simultaneous calls between all three of them. Since then I have had only constant problems, mainly that after 3 calls on a phone, if they try to xfer or do any ohter things (sometimes just answer the 4th call) the phone freezes, they have had this happen to them throughout the week. Until yesterday I decided it must be a frimware problem, so I downgraded them to 7.1. Since then (around 5PM EST yesteday) it didn't happen *yet*. So I'm assuming it has to do with the firmware. So my question is, is anybody else using 7.5 firmware? If yes, do you have all the line buttons configured to the same SIP account? If yes, do you see the same problem? I also noticed that with 7.5 firmware callwaiting has to be enabled for the second call to be able to come in, otherwise the phone returns a Busy here, while with the older versions it could have been disabled and it worked fine, the phone only returned busy here on the 7th call. So I had to enable call waiting, the way I did it was that in the SIP.cfg file I added call_waiting: 3 I'm not sure if this is related or not, but that was the only change I had to do to the config files. It's nothing you did... I did the same thing. Went from 7.4 to 7.5 and all sorts of weird things started happening. The biggest of which was lines 2-6 wouldn't register or display the same busy message you got. I also got double ringing which someone told me how to fix. The phones locked up... I rolled back to 7.4 and have had no issues since. Good luck! Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP601 Question
Hey guys, Thanks for the suggestions. I did find the problem. Looking in the sip debug, I was getting a 407 error, corrected that, then was getting a 404. Which lead me to look at my context and bam...typo, I had conext instead of context. Corrected that and all is well. Thanks again. Kevin C F wrote: What does the dialplan for the Polyocm 601 (the one the phone uses, not Asterisk) look like? You can see if it's a polycom or asterisk thing, by enabling sip debug, and watch what is coming in from the Polycom. if nothing is coming then it's the Polycom doing it. On 2/23/06, Kevin Smith <[EMAIL PROTECTED]> wrote: Hey everyone, I haven't seen an issue quite like mine, so I am hoping anyone who used the Polycom 601's may have an idea. We are going to be switching our office over to Asterisk. All the phones are going to be 601's, I am going to set up a boot server, but for now I am just going to test everything on one phone. My question is I have the phone registered in Asterisk (phone icon on the polycom is black), but I cannot make any calls. I tried to dial the extension shown in the extensions.conf file and I just get a busy signal. However, if I plug in an old budgetone 100 with the same settings, it works just fine. Any ideas? Also, when setting up a boot sever, the phone updates the log entries and there is a cfg file for the mac address of the phone, but during a reboot it cannot connect to the bootserver. Do I need to have the sip.ld, etc, files up there for it to work properly? Any suggestions would be greatly appreciated. Here is the sip.conf file [test] type=friend secret=blahpoly insecure=yes host=dynamic qualify=500 nat=no mailbox=testmailbox callerid=Yourname conext=local disallow=all allow=ulaw progressinband=no here is the local section of the dial plan. exten => 850,1,Goto(Mercury-Network,850,1) exten => 888,1,VoiceMailMain(@Mercury-Network-Emp) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP601 Question
> Hey everyone, I haven't seen an issue quite like mine, so I am hoping
> anyone who used the Polycom 601's may have an idea.
>
> We are going to be switching our office over to Asterisk. All the phones
> are going to be 601's, I am going to set up a boot server, but for now I
> am just going to test everything on one phone. My question is I have the
> phone registered in Asterisk (phone icon on the polycom is black), but I
> cannot make any calls. I tried to dial the extension shown in the
> extensions.conf file and I just get a busy signal. However, if I plug in
> an old budgetone 100 with the same settings, it works just fine. Any ideas?
>
> Also, when setting up a boot sever, the phone updates the log entries
> and there is a cfg file for the mac address of the phone, but during a
> reboot it cannot connect to the bootserver. Do I need to have the
> sip.ld, etc, files up there for it to work properly?
>
> Any suggestions would be greatly appreciated.
>
> Here is the sip.conf file
>
> [test]
> type=friend
> secret=blahpoly
> insecure=yes
> host=dynamic
> qualify=500
> nat=no
> mailbox=testmailbox
> callerid=Yourname
> conext=local
> disallow=all
> allow=ulaw
> progressinband=no
>
> here is the local section of the dial plan.
> exten => 850,1,Goto(Mercury-Network,850,1)
> exten => 888,1,VoiceMailMain(@Mercury-Network-Emp)
Try this instead:
[850] ; [test]
type=friend
username=850<- add this line
secret=blahpoly
; insecure=yes <- no need for this
host=dynamic
; qualify=500 <- no need for this now
nat=no
mailbox=850 ; testmailbox
; callerid=Yourname <- no need for this now
conext=local
disallow=all
allow=ulaw
; progressinband=no <- no need for this
dtmfmode=rfc2833 <- add this line
In the extensions.conf dialplan, include something like this:
[local]
exten => 850,1,Dial(SIP/850,15)
exten => 850,2,Voicemail(u850)
exten => 850,102,Voicemail(b850)
exten => 850,103,Hangup
; Voicemail access (prompts for exten and password)
exten => 888,1,Wait,1
exten => 888,2,VoicemailMain
exten => 888,3,Hangup
; Voicemail access (does not prompt for anything)
exten => 889,1,Wait,1
exten => 889,2,VoicemailMain(s${CALLERID(num)})
exten => 889,3,Hangup
Configure one of the 601's with matching username and secret, and
get it to register with asterisk. If 'sip show peers' does not reflect
the IP address of the 601, it ain't registered. Work on correct that
first.
___
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RE: [Asterisk-Users] What business IP phone to use
> Simple formula: > > 1. Total Revenue > 2. % of revenue derived from phone usage > 3. =Cost of downtime by using SoHo or consumer gear. > > It's not a question of if a SoHo or low cost device will screw up, it is a > question of when. This is 23 years of experience talking. > > Where I work, the value of #3 above is $16 Cdn a *second*. We are below 500 > employees, so we fall into the SMB segment. Sometimes I'm appalled by > statements that a $700 switch or a $400 phone isn't worth it. Huh?? Maybe in Absolutely right! for something as critical as switches & cabling I always recommend to spend real money. Don't ever try to save money any equipment that is required to operate the business. (Had very good experience with HP procurves over the last 10 years or so). There is no point buying netgear or other low-cost switches for a business ever. The cost saving of being able to pin-point a cabling/NIC/bandwidth problem down to the port on the switch easily and quickly is wonderful. Combined with SNMP and all the other goodies good switches come with, our clients save a lot of money by paying me less for my time ( d'oh ;-) ). The difference can also cause unnecessary delays and therefor echo in the path. For example, procurve switches typically have 13ms switching time, the high-end netgears about 21ms. As soon as you stack a couple of switches you are talking 26ms vs 42ms extra delay in the path! I see no reason however to spend $400 on a single phone though, because if a single phone breaks, it's not going to bring your business to a standstill, is it? (I guess unless you only have one in the first place ;-) ) conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems
On 2/23/06, C F <[EMAIL PROTECTED]> wrote: So my question is, is anybody else using 7.5 firmware?I haven't had the same issue you describe so I can't help, sorry. But I have had other issues with 7.5 firmware. If asterisk restarts then the phone needs to be rebooted to re-register. I haven't seen anything online that describes it except this: http://groups.google.com/group/Asterisk-users/browse_thread/thread/a0e4848fff557dd/13221440af999173?q=7.5+firmware+restart&rnum=1#13221440af999173hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Explain Yellow Alarm in a Legacy Integration
How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration?We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P.Twice today (first time in over a month) we received a Yellow Alarm on the TE110P. I have been able to clear it easily by restarting zaptel. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems
Thanks. Makes me feel good to see Cisco goofing :) On 2/23/06, Jeremy Koski <[EMAIL PROTECTED]> wrote: > > We had all kinds of problems with SIP 7.5, and decided to stay with 7.4. > > We had phones freeze, reboot, and display XML parse error until we removed > 7.5. We also had a lot of dropped calls. > > > > > > On Thu, 23 Feb 2006, C F wrote: > > > I recently updated my phones Cisco 7960 phones (3 of them) in a high > > volume call place, where the Secretaries use the 7960 phones to answer > > inbound calls, as many as 15 simultaneous calls between all three of > > them. > > Since then I have had only constant problems, mainly that after 3 > > calls on a phone, if they try to xfer or do any ohter things > > (sometimes just answer the 4th call) the phone freezes, they have had > > this happen to them throughout the week. Until yesterday I decided it > > must be a frimware problem, so I downgraded them to 7.1. Since then > > (around 5PM EST yesteday) it didn't happen *yet*. So I'm assuming it > > has to do with the firmware. > > So my question is, is anybody else using 7.5 firmware? > > If yes, do you have all the line buttons configured to the same SIP account? > > If yes, do you see the same problem? > > > > I also noticed that with 7.5 firmware callwaiting has to be enabled > > for the second call to be able to come in, otherwise the phone returns > > a Busy here, while with the older versions it could have been disabled > > and it worked fine, the phone only returned busy here on the 7th call. > > So I had to enable call waiting, the way I did it was that in the > > SIP.cfg file I added > > call_waiting: 3 > > I'm not sure if this is related or not, but that was the only change I > > had to do to the config files. > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
> Ah! Now this is actually something I've not been able to get my head around: > > > Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which > > has its own MP3 player. > > Can anybody tell me where this built-in MP3 player is in 1.2.x/how do I > use it ? > > I still seem to have the usual two mpg123 processes running with 1.2.4, > with whatever music on hold is set in musiconhold.conf > > I'm sure it is very obvious, but I can't for the life of me figure out > what I'm supposed to do to use the built-in MP3 player facilities. > > > I just have the following in my musiconhold.conf: > > [default] > mode=mp3 > directory=/var/lib/asterisk/mohmp3 > random=yes Since I just went through at least a part of the above, I think the following might be an approach to consider. Download 'asterisk-addons' (if you don't already have it). cd /usr/src/asterisk-addons/format_mp3 and read the README file. The 'make clean install' Then, in /etc/asterisk/musiconhold.conf, [default] ; mode=quietmp3 mode=files directory=/var/lib/asterisk/mohmp3 Stop and restart asterisk. mpg123 is no longer in use. The above works just fine with the asterisk-provided musiconhold files located in the above mohmp3 directory. If you add other mp3 files into that directory, you might have to convert them per the README. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TFTP server for GrandStream BT phones / need testing
> This patch adds only GS BT phones recognition funcionality. > tftpd-hpa does not correct handle the OPTION parameters in the TFTP packet ;( > At first I tired to implement it into tftpd-hpa, but after debuging the code > I give it up. The tftpd-hpa reads the parameters from the TFTP packet as they > are in the > packet and after each option it do the associated command. > > The packet from GS BT looks like > --- cut --- > Opcode: Read Request (1) > Source File: boot.bin > Type: octet > Option: blksize = 1024 > Option: tsize = 0 > Option: timeout = 4 > Option: grandstream_MODEL = BT-100 > Option: grandstream_NAT = 1 > Option: grandstream_ID = 000b8203e0e9 > Option: grandstream_REV_BOOT = 001.000.001.000 > Option: grandstream_REV_PHONE = 001.000.006.007 > Option: grandstream_REV_VOC = 001.000.001.000 > Option: grandstream_REV_HTML = 001.000.000.049 > Option: grandstream_REV_RING1 = 001.000.000.000 > Option: grandstream_REV_RING2 = 001.000.000.000 > Option: grandstream_REV_RING3 = 000.000.000.000 > --- cut --- > > So it reads ... > Opcoce - do something > Source File - send the file to the client ... > Reads the Options from packet. So in the time or sending requested file, > there I have no information about the > phone. I was too lazy to correct this ;( > oh I see, that would make the provisioning work through routers ;-) I use the MAC address to match different phones, which obviously breaks if I need a router between phone and tftpd-server. Thanks for clarifying and well done! Conrad > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: VoIP over bonded link
>It's stupid. Don't ever connect 2 different building with copper. >Just wait until you get some kind of lightening hit or electrical >fault, but make sure you are no where near it. Use fibre. Thanks for the reply. Unfortunately, the conduit for the provisioning of the new building is unsuitable for fibre (too many sharp bends) and we can't core out the concrete and put in a new conduit because of obstacles in the way that make laying new conduit impractical, so we are stuck with (existing) copper. We already have copper-to-copper connections of different types (electrical, security etc) between the buildings so a lightning strike is going to hose us no matter what. That aside, does anyone have opinions on my original question as to the suitability of bonded links for VoIP? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Cisco 79xx and SIP 7.5 Problems
Try 7.4 (P0S3-07-4-44). Notice on voip-info.org the comments about 7.5 being buggy with some sort of registration issues (if I recall). Also, voip-info reports 7.4 to be stable. If you read the Cisco changelog, you'll see LOTS of bug fixes up to 7.4 (no new features, just fixes). I THINK one of the bug fixes was that if call-waiting was disabled, it had no effect, and still used call-waiting (as you correctly report). Good luck! >I recently updated my phones Cisco 7960 phones (3 of them) in a high volume call place, where the Secretaries use >the 7960 phones to answer inbound calls, as many as 15 simultaneous calls between all three of them. >Since then I have had only constant problems, mainly that after 3 calls on a phone, if they try to xfer or do any >ohter things (sometimes just answer the 4th call) the phone freezes, they have had this happen to them throughout >the week. Until yesterday I decided it must be a frimware problem, so I downgraded them to 7.1. Since then (around >5PM EST yesteday) it didn't happen *yet*. So I'm assuming it has to do with the firmware. >So my question is, is anybody else using 7.5 firmware?. >If yes, do you have all the line buttons configured to the same SIP account? >If yes, do you see the same problem? > >I also noticed that with 7.5 firmware callwaiting has to be enabled for the second call to be able to come in, otherwise the phone returns a Busy here, while with the older versions it could have been disabled and it worked fine, the phone only returned busy here on the 7th call. So I had to enable call waiting, the way I did it was that in the SIP.cfg file I added call_waiting: 3 >I'm not sure if this is related or not, but that was the only change I had to do to the config files. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems
We use mainly 7940's in our environment, I currently have about 60 of them on my system, and the 7.5 firmware really screwed up our phone network. Not in the same way yours has, but there seem to be a lot of glitches, even jumping from 7.4 to 7.5. Have you tried the 7.4 firmware to see if that does you any good? Aaron C F wrote: I recently updated my phones Cisco 7960 phones (3 of them) in a high volume call place, where the Secretaries use the 7960 phones to answer inbound calls, as many as 15 simultaneous calls between all three of them. Since then I have had only constant problems, mainly that after 3 calls on a phone, if they try to xfer or do any ohter things (sometimes just answer the 4th call) the phone freezes, they have had this happen to them throughout the week. Until yesterday I decided it must be a frimware problem, so I downgraded them to 7.1. Since then (around 5PM EST yesteday) it didn't happen *yet*. So I'm assuming it has to do with the firmware. So my question is, is anybody else using 7.5 firmware? If yes, do you have all the line buttons configured to the same SIP account? If yes, do you see the same problem? I also noticed that with 7.5 firmware callwaiting has to be enabled for the second call to be able to come in, otherwise the phone returns a Busy here, while with the older versions it could have been disabled and it worked fine, the phone only returned busy here on the 7th call. So I had to enable call waiting, the way I did it was that in the SIP.cfg file I added call_waiting: 3 I'm not sure if this is related or not, but that was the only change I had to do to the config files. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
Hi, I still seem to have the usual two mpg123 processes running with 1.2.4, with whatever music on hold is set in musiconhold.conf I'm sure it is very obvious, but I can't for the life of me figure out what I'm supposed to do to use the built-in MP3 player facilities. [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 random=yes Change mode=mp3 to mode=files then do a "moh reload" on CLI -- Joel Vandal ScopServ Inc. http://www.scopserv.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup Detection
Is it possible to patch the zaptel drivers (or whatever appropriate files) to use DTMF tone "D" for hangup detection? I have a Toshiba PBX which does not provide CPC by any means other than congestion or D tone. Thanks, Aaron Picht ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx and SIP 7.5 Problems
We had all kinds of problems with SIP 7.5, and decided to stay with 7.4. We had phones freeze, reboot, and display XML parse error until we removed 7.5. We also had a lot of dropped calls. On Thu, 23 Feb 2006, C F wrote: I recently updated my phones Cisco 7960 phones (3 of them) in a high volume call place, where the Secretaries use the 7960 phones to answer inbound calls, as many as 15 simultaneous calls between all three of them. Since then I have had only constant problems, mainly that after 3 calls on a phone, if they try to xfer or do any ohter things (sometimes just answer the 4th call) the phone freezes, they have had this happen to them throughout the week. Until yesterday I decided it must be a frimware problem, so I downgraded them to 7.1. Since then (around 5PM EST yesteday) it didn't happen *yet*. So I'm assuming it has to do with the firmware. So my question is, is anybody else using 7.5 firmware? If yes, do you have all the line buttons configured to the same SIP account? If yes, do you see the same problem? I also noticed that with 7.5 firmware callwaiting has to be enabled for the second call to be able to come in, otherwise the phone returns a Busy here, while with the older versions it could have been disabled and it worked fine, the phone only returned busy here on the 7th call. So I had to enable call waiting, the way I did it was that in the SIP.cfg file I added call_waiting: 3 I'm not sure if this is related or not, but that was the only change I had to do to the config files. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fromstring when sending e-mail on recievedvoicemail
Hello Arne, Arne Morten Johansen, 21.02.2006 (d.m.y): > Just one more question. In /etc/passwd there's a line with "asterisk" > and "added by portage" in it. Can I just change this without screwing > up everything? You can change these entries except from the number at the beginning of the line. > Or is there a command to change user info or something? > As you can see, I'm not so good in Linux. This number I mentioned before is the numerical User ID. While human beings deal with names the (UNIX-like) system only deals with numbers and identifies users by their numerical UIDs. Regards, Christian -- Vorläufig habe ich noch keine Lust ins bessere Jenseits zu beißen. -- Heinz Erhardt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys WIP300 WiFi Phone
Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit didn't come with a CD-ROM or registration card, eventhough they are listed as being in the Package Contents. - This phone uses a USB port to charge, do firmware updates, and perhaps other things. Sadly... it DOES NOT COME WITH A USB CORD! You'd think for $250 that you'd get a cord included... oh well. It does come with a charger with a usb end so you can charge the phone from an AC outlet, though. - The battery charging animation runs backwards, animating like the battery's charge is flowing out rather than in. A little amusing. No charge status while it is charging, which I don't like. It would be nice to see that it's, say, 75% charged for example. Does anyone else have one of these phones yet? Any gotcha's as far as using it with *? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] isdn problem
Hi I have beronet BN8S0 isdn card in my asterisk and , card is working fine, but when I try to dial to special number 118913 ( telephone number information) from polish telecom TPSA, I always geting timeout . Bellow is isdn signaling dump : --> * CallGrp: PickupGrp: --> rxgain:0 txgain:0 --> * dad:118913 tech:mISDN/2-u25 ctx:default --> * Setting Context to from-tpnet --> TON: Unknown --> TON: Unknown --> PRES: Allowed (0x0) --> SCREEN: Unscreened (0x0) --> * adding2newbc ext 118913 --> * adding2newbc callerid 717201234 I SEND:SETUP oad:717201234 dad:118913 port:2 --> mode:TE cause:16 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:0 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:0 addr:51400102 l3id:11000c --> new_l3id 11000e --> * SEND: State Dialing pid:43 I IND :SETUP_ACKNOWLEDGE oad:717201234 dad:118913 port:2 --> mode:TE cause:16 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:1 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:43 addr:51400102 l3id:11000e I IND :TIMEOUT oad:717201234 dad:118913 port:2 --> mode:TE cause:16 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:1 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:43 addr:51400102 l3id:11000e I IND :RELEASE oad:717201234 dad:118913 port:2 --> mode:TE cause:-1 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:1 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:43 addr:51400102 l3id:11000e Idx:0 stack->cchan:0 Chan:1 Idx:1 stack->cchan:0 Chan:2 Idx:0 stack->cchan:0 Chan:1 Idx:1 stack->cchan:0 Chan:2 I IND :CLEAN_UP oad: dad: port:2 --> mode:TE cause:16 ocause:16 rad: --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 --> channel:0 caps:Speech pi:0 keypad: --> urate:0 rate:0 mode:0 user1:0 --> pid:0 addr:51400102 l3id:11000e Trying to Release bc with l3id: 11000e * RELEASING CHANNEL pid:0 ctx:from-tpnet dad:118913 oad:118913 state: (null) --> * State Down --> Setting AST State to down * --> In State Dialin * --> Queue Hangup What I've tried to do: - set correct CallerID - use Dial app with option 's', 'n:h' , 'h' Without luck, but when I connect analog telephone to NT R-interface , after dialing number I have connection. Other thing is there is second similar number 118912 (abroad telephone number information ) , and I call to this number without problems. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Okay can somebody explain this...
Hi, I got rid of the messages I was getting in the CDR (pbx.c: Cannot find extension context 'default') by adding a blank 'default' context at the front of my extensions.conf (I use the context 'extensions-home') this also (well sort of ) fixed my problem with blind transfers. I can blind transfer using '##' (set in features.conf) once but I cannot transfer a second time (I was unable to transfer at all before adding the blank 'default' context)?? Now I am really confused. Do I have to have a context called 'default' for this to work properly? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor a call in progress.
Steve Totaro wrote: 1. go to www.google.com 2. type "asterisk monitor application" 3. click on the first result 4. read and implement 5. google is your friend I hope I made myself clear too ;-P Hi Steve, I made it before and found the same problem. Are you referencing to this code? exten => 2060,1,Answer exten => 2060,2,Wait(1) exten => 2060,3,Monitor(wav,myfilename) exten => 2060,4,Meetme(1,ps) But it implies in receiving a call and start monitoring it. I intent to create this behavior. A client call. A user answer. Another user, a manager, for instance. Dial a code: For instance: exten => 1010,1,() #Start to listen the call placed in the channel 1 exten => 1011,1,() #Start to listen the call placed in the channel 2 And so on... Fernando Lujan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Streaming Music On Hold - Reality Check
Yes, it streams forever. If you start Asterisk, run an ngrep command or other
monitoring tool. You will see that as soon as you start Asterisk, it starts to
receive a stream. If it didn't do this, then when someone went on hold, it
would have to connect, which may take a number of seconds.
-Original Message-
From: Doug Geary [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Streaming Music On Hold - Reality Check
Thanks to this thread, we got it working too... but have a question...
Once this is setup... does it stream forever, or does the stream only
start when someone goes on hold/into a queue/etc?
If it streams forever, at 24k... it looks like over 7GB/month in
bandwidth... so we're not going to want to do that if a) it streams
constantly and b) my math is correct.
Thanks,
Doug
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Douglas Garstang
> Sent: Wednesday, February 22, 2006 4:18 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
> Commercial Discussion
> Subject: RE: [Asterisk-Users] Streaming Music On Hold
>
> Thanks. I got it working. Yay.
>
> Now, it seems that Asterisk is very fussy with the streams. A lot
don't
> work, especially when the URL ends in something.pls. Anyone know if
that's
> true? Is Asterisk's support of this still pretty limited?
>
> Doug.
>
> -Original Message-
> From: Jonathan Augenstine [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, February 22, 2006 10:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Streaming Music On Hold
>
>
> Try this:
>
> musiconhold.conf:
>
> [stream2]
> mode=mp3
> directory=http://pubint.ic.llnwd.net/stream/pubint_wnpr
>
>
> extensions.conf:
>
> exten => 1234,1,Answer
> exten => 1234,2,MusicOnHold(stream2)
> exten => 1234,3,Hangup
>
>
> On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote:
> > Ok, I'm tearing my hair out trying to get Asterisk moh streaming to
> work. After several hours jerking around with icecast and muse, I
tried to
> point my asterisk system directly at two streams I know work.
> >
> > This is what extensions.conf has:
> >
> > [default]
> > mode=quietmp3
> > directory=/var/lib/asterisk/mohmp3
> >
> > [stream2]
> > mode=custom
> > directory=/var/lib/asterisk/mohmp3-empty
> > application=http://pubint.ic.llnwd.net/stream/pubint_wnpr
> >
> > and this is how I am testing it:
> > exten => 1234,1,Answer
> > exten => 1234,2,SetMusiconHold(stream2)
> > exten => 1234,3,WaitmusiconHold(60)
> > exten => 1234,4,Hangup
> >
> > and this is the console output I get when I dial 1234:
> >
> > Asterisk Ready.
> > *CLI> -- Executing Answer("SIP/3250072-ed28", "") in new stack
> > -- Executing SetMusicOnHold("SIP/3250072-ed28", "stream2") in
new
> stack
> > -- Executing WaitMusicOnHold("SIP/3250072-ed28", "60") in new
stack
> > -- Started music on hold, class 'stream2', on channel
'SIP/3250072-
> ed28'
> > -- Stopped music on hold on SIP/3250072-ed28
> >
> > If I replace SetMusiconHold(stream2) with SetMusiconHold(default), I
get
> the default music on hold. Running ngrep on port 80 shows me that the
> Asterisk system is not sending or receiving ANY data on port 80. What
am I
> doing wrong? Yes, it has network and DNS connectivity.
> >
> > Can't believe it's this hard!
> >
> > Doug.
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] OT: VoIP over bonded link
> I have to provision several dozen * users to a seperate building on our > campus in the same subnet. Ordinarily, I'd just run a gigabit cat6 cable to > another switch if it doesn't violate the 100 metre rule, but this building > is several hundred metres away from my backbone. My only option for cabling > to the remote building is copper. My plan is to provision them with a Linux > bridge with 4 NIC's: 1 gigabit to the backbone, and three bonded together as > a single interface (90 mbit aggregate), then plugged into this dealie: > > http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946 > > At the remote building, the reverse: another Linux box with 4 NIC's that > de-aggregates the link to a gigabit connection on a switch, and then to the > wall plates. I'm pretty sure this will work for data no problem, but I'm a > little concerned about latency on a timing-sensitive applicaiton like VoIP. > > Anyone have experience with VoIP over bonded link? Is there a gotcha? Is > this a stupid idea? On my whiteboard it looks fine! Some thoughts that you might want to consider... The vdsl box runs at speeds "up to 15 meg". That translates into the longer the copper loop, the slower the speed. You'll probably want to accurately measure the copper loop length and translate that into some 'expected' speed. Probably won't be 15 meg, and whatever the documentation suggests, it will likely be a fair amount slower then that. Does the vdsl truly operate in a full duplex mode with equal bandwidth in either direction? We've worked with many corporations and institutions in over 40 states doing network performance assessments, and seldom (if ever) do bonded interfaces actually work the way that you might think they work. I've not spent any time with the linux bonding that you're considering, but you might want to better understand exactly how that works. E.g., some bonding actually functions at 'per packet' level, which implies the maximum speed of any single packet is the speed of one vdsl circuit. If one of the bonded circuits has errors, what impact does it have on the other three error-free circuits. (Its not uncommon for one interface to have very significant impact on all other interfaces.) If all of the above can be answered with positive thoughts, you'll still want to consider some form of QoS on those links to ensure the voip packets are not held in a queue. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail problems
Have you recorded an unvailable and/or busy message? On 2/23/06, Michaël Gaudette <[EMAIL PROTECTED]> wrote: > > Hi, > > I've asked this question in the past, but I didn't get a precise answer. > Hopefully somebody will take note of my question. > Before I forget, I am using Asterisk 1.2.4. > > I've been using the Voicemail app with success (i.e. it works) except for > one single thing: the ONLY message that it ever played back to the caller is > the temporary message. If I delete the temporary message (through the > voicemail menu), then a generic message is played (not one that I've > recorded myself). > > I am not sure if the unavailable message or the busy message should be > played, but neither are. My .conf file is this: > > exten => 1,1,Dial(SIP/grandstream2000|20) ; 20 seconds of > trying my GrandStream GXP2000 > exten => 1,2,VoiceMail([EMAIL PROTECTED]) > > What could be the issue? > > Mike > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
Soner Tari wrote: I vote for the raw file format, due to the reasons listed here: http://www.orderlyq.com/asteriskqueues.html Of course you need to convert all mp3 moh files to raw format manually, but it's easy as described there. We were using the rawplayer method on our server, but it ended up spawning hundreds of zombie processes. I talked to Kevin Fleming about it, and he recommended switching to native MOH. Scalability is a big concern of mine, so I asked him about the impact it would have on CPU, memory, and disk utilization. His response was, "not much, more memory usage though." I could spare the memory, so I gave it a shot. So far, I'm very impressed. It eliminated the zombie processes completely and the quality of the playback is *much* better. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
It happened with g729a as well -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Thursday, February 23, 2006 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning On Feb 23, 2006, at 4:58 AM, Adam Robins wrote: > Thanks, > > We already have a cron reboot of all of our Asterisk servers every > night. We've been doing this for over a year due to memory leak > issues. ??? What do you think this is windows 95??? I had a problem like that I would be looking at getting rid of asterisk. I don't ;~) I wonder what your leak is ? > > After 2 weeks of messing around with every conceivable IAX2 and > jitterbuffer configuration, I switched to SIP yesterday. Complaints > went from 10-20 per day to ZERO. Literally overnight. I wonder if this is an ILBC frame size issue of some sort? Seems odd. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended rack-mountable server anyone?
Hi, we have used about 6 Intel rack mounted servers, with dual Xeon processors, (I have forgotten the exact make.) we used them with Digium quad span pri cards, and some with Sangoma cards, works well . These servers are still pretty solid. they come with two onboard NICs, and with the Digium cards, one of the NICS had to be disabled cause of interrupt clashes. Other wise they have worked well. > I forgot about one other issue we had with the 2850. The integrated > NICs caused interrupt issues with the TE411P. We had to disable the > integrated NICs, and installed dual port gigabit intel NIC. > > Stagg Shelton > www.oneringnetworks.com > > [EMAIL PROTECTED] wrote: > >>Alexander, Perhaps I'm wrong, but I have a server here next to my desk >>(IBM e325) and I tried to fit a normal pci card into it. The slots are >>completely different and the card would not fit.. this was just a pci >>dvi video card. The server specs say that it is using PCI-X technology >>for the slots so this leads me to believe that they are not compatible >>as one would think. >> >>Cory Andrew, I will look into the supermicro servers again, I'm not >>keen on the handles up front on them though, that makes for awkward >>handling (imo). >> >>Wow Stagg, Thank you for that first hand knowledge. These are things >>you just can't learn until you buy a product and experience it first >>hand. I'm not so sure that we want to Frankenstein our own cable for >>this configuration though! (yikes!) >> >>Hopefully some other people will pipe up too with some more server >> suggestions! >> >>On 2/22/06, Stagg Shelton <[EMAIL PROTECTED]> wrote: >> >> >>>We just installed asterisk for a customer using a Dell 2850. It has 3 >>>pci slots. My customers configuration contained a TE411p Quad Span PRI, >>>and a TDM400P with 4 FXS Modules. The only problem that we had with the >>>2850 was getting power to the TDM400P. We located a power connector on >>>the backplane that supplied the required 12v. I think it was originally >>>intended to power a tape backup drive. We ultimately sacraficed a power >>>supply to get at it's 12V P4 connector. We then used a voltmeter to put >>>together a pinout for the dell power port, and frankensteined together a >>>cable that could be used to power the TDM400P.Aside from the power >>>issue, the platform seems rock solid. >>> >>>Stagg Shelton >>>www.oneringnetworks.com >>> >>> >>>[EMAIL PROTECTED] wrote: >>> >>> >>> Hey everyone, I've been doing a lot of research into a decent server for Asterisk but I seem to be running and circles and now I am turning to you. The issue I have is it needs to be rack mountable (so a Dell SC430 isn't going to work) and preferably have 3 pci ports. The problem that I seem to be running into is that when I look at servers from Dell or IBM or the like they only seem to support PCI-X which (from what I understand) does not support the Digium cards that we already have and that they still make. So if anyone has a suggestion or has a server they rather prefer for it's reliability, expandability, etc, please recommend it! Thank you in advance, Mitchel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >>>___ >>>--Bandwidth and Colocation provided by Easynews.com -- >>> >>>Asterisk-Users mailing list >>>To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>___ >>--Bandwidth and Colocation provided by Easynews.com -- >> >>Asterisk-Users mailing list >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail problems
> I've asked this question in the past, but I didn't get a precise answer.
> Hopefully somebody
will take note of my question.
> Before I forget, I am using Asterisk 1.2.4.
>
> I've been using the Voicemail app with success (i.e. it works) except for one
> single thing: the
ONLY message that it ever played back to the caller
> is the temporary message. If I delete the temporary message (through the
> voicemail menu), then
a generic message is played (not one that I've
> recorded myself).
>
> I am not sure if the unavailable message or the busy message should be
> played, but neither are.
My .conf file is this:
>
> exten => 1,1,Dial(SIP/grandstream2000|20) ; 20 seconds of trying my
> GrandStream GXP2000
> exten => 1,2,VoiceMail([EMAIL PROTECTED])
>
> What could be the issue?
Not sure what you are really doing with the above, but look over the
following examples.
To dial extension 3000, one would do something like this:
exten => 3000,1,Dial(SIP/3000,15)
exten => 3000,2,Voicemail(3000|ug(6))
exten => 3000,102,Voicemail(3000|bg(6))
exten => 3000,103,Hangup
For a working extension, to check voicemail do something like this in
your extensions.conf:
; Voicemail access (prompts for exten and password)
exten => 3998,1,Wait,1
exten => 3998,2,VoicemailMain
exten => 3998,3,Hangup
; Voicemail access (does not prompt for anything)
exten => 3999,1,Wait,1
exten => 3999,2,VoicemailMain(s${CALLERID(num)})
exten => 3999,3,Hangup
And be sure that voicemail.conf has an entry like this:
3000 => 3000,Your Name,your-email-address
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RE: [Asterisk-Users] Best ATA for general residential deployment??
What a load of @#$%! The 102 is obviously a ripoff of the HandyTone 486! Even the spec sheet is copied! Like HT486 only much better? I doubt it. I have it on good authority that Grandstream is taking legal and police action against these people and if there is any justice in the world they will be out of business soon. I'm all for competition, but piracy is another thing altogether. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Tele Cost Price Reducer [mailto:[EMAIL PROTECTED] Sent: Thursday, February 23, 2006 9:12 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionCc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Best ATA for general residential deployment?? hi , i have some options we are working with at vast deployement with no problems: www.tigernetcom.com - type 102 is a nice ATA, like GS 486 but far away better. we import directly from the producer at great prices so if anybody interested, please contact off-list. additionaly, we know another excellent producer (price of about 45$ FOB china) we would recomend off-list. Mickey On 2/23/06, Marc Rys <[EMAIL PROTECTED]> wrote: My $.02 is that HT486 sucks as a router. It works well as a ATA. I have 5 and have all of them are behind separate routers. The HT486 never gave me the full download speed of my cable modem and even when my PC wasn't powered up, and I wasn't talking on the HT486 my cable modem still looked like the HT486 was sending traffic non-stop. I put a Buffalo router in front of the HT486 and all is good. Cable modem doesn't look full of activity during idle time and my PC can use the cable modem to fullest potential. It's a shame too, because I bought the HT486's for the router capability. It turned out to be a waste.Marc-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of The VoIP ConnectionSent: Wednesday, February 22, 2006 5:11 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Best ATA for general residential deployment??Absolutely. HT-486 is my pick for best all-around unit based on ease-of-use, value, performance and reliability. -MikeMichael CrownManaging Partnerwww.thevoipconnection.com321.989.6728 ext. 611sip:[EMAIL PROTECTED]> -Original Message-> From: Martin Joseph [mailto:[EMAIL PROTECTED]]> Sent: Wednesday, February 22, 2006 2:10 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: Re: [Asterisk-Users] Best ATA for general> residential deployment??>>> On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote: >> > On 2/22/06, Matt <[EMAIL PROTECTED]> wrote:> >> Yes.. there are provisioning tools that you have to get.> >> Unfortunately it's this catch 22 loop. You have to prove that you > >> can offer 200+ ATAs to customers, or you can't get the tools, but> >> yet, you don't really want to offer those ATAs to the customer's> >> without having the tools.> > > > This sounds like yet another reason to avoid purchasing Sipura> > equipment and supporting Sipura in any way. I don't know about you> > guys, but I have better things to do than screw around with asinine > > vendor policies that make it more difficult than necessary to get> > things done.> >> True, but it's kind of a "pick your poison" situation in my opinion.> Ht-486 anyone? >>>___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.375 / Virus Database: 268.0.0/266 - Release Date: 2/21/2006--No virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.375 / Virus Database: 268.0.0/266 - Release Date: 2/21/2006 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?
Hi, Is setting the variable _TRANSFER_CONTEXT required in features.conf for Asterisk 1.2.4? It appears from the Wiki that transfers across contexts are not possible when this is set. If it is not set can one trasfer across contexts?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: VoIP over bonded link
On Thursday 23 Feb 2006 17:30, Colin Anderson wrote: > I have to provision several dozen * users to a seperate building on our > campus in the same subnet. Ordinarily, I'd just run a gigabit cat6 cable to > another switch if it doesn't violate the 100 metre rule, but this building > is several hundred metres away from my backbone. My only option for cabling > to the remote building is copper. My plan is to provision them with a Linux > bridge with 4 NIC's: 1 gigabit to the backbone, and three bonded together > as a single interface (90 mbit aggregate), then plugged into this dealie: > > http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946 > > At the remote building, the reverse: another Linux box with 4 NIC's that > de-aggregates the link to a gigabit connection on a switch, and then to the > wall plates. I'm pretty sure this will work for data no problem, but I'm a > little concerned about latency on a timing-sensitive applicaiton like VoIP. > > Anyone have experience with VoIP over bonded link? Is there a gotcha? Is > this a stupid idea? On my whiteboard it looks fine! It's stupid. Don't ever connect 2 different building with copper. Just wait until you get some kind of lightening hit or electrical fault, but make sure you are no where near it. Use fibre. B -- http://www.mailtrap.org.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom boot times/XML files.
On 2/23/06, Anton Krall <[EMAIL PROTECTED]> wrote: > Ken, Im having problems with the time on my polycoms, it doesn't matter > which sntp server and offset I enter, the phone wont take the offset into > account, Ive tried entering it directly on the phone and also on the .cfg > file but no luck, any tips? I had this happen too. Nothing I did fixed the time and every reboot takes a while as I've whined about here before. Resetting user parameters and an ftp reprovision did the trick. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 601 Buddy Watch problems
On Thu, 23 Feb 2006, Nathan Bowyer wrote: It may have something to do with the watch limit on the Polycom firmware. I have one phone that does this as well, nearly all the time. I believe its the only one that exceeds the 6-7 buddy watch limit as well. Now that polycom is "committed to working with asterisk", is polycom going to fix the 7 buddy watch limit? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] not consistent log from asterisk
Hello,
I have 2 channels in iax.conf
[iaxfwd]
type=user
callerid= Free World Dialup
inkeys=freeworlddialup
auth=rsa
context=incoming
qualify=yes
[iaxfwd-outbound]
type=peer
host=iax2.fwdnet.net
username=xx
secret=***
auth=md5
The problem is:
When I tell FWD to call me I have this output in my asterisk
consol:
Executing Dial("IAX2/iaxfwd-outbound-3", "SIP/|PHONE_1|30")
in new stack
If I comment iaxfwd-outbound channel [iaxfwd-outbound],
then the output is correct:
Executing Dial("IAX2/192.246.69.186:4569-1",
"SIP/PHONE_1|30") in new stack.
(192.246.69.186:4569 : this is from FWD)
Thank's for any help
a+
--
Bayrouni
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Re: [Asterisk-Users] Best ATA for general residential deployment??
On Feb 23, 2006, at 6:11 AM, Tele Cost Price Reducer wrote: hi , i have some options we are working with at vast deployement with no problems: www.tigernetcom.com - type 102 is a nice ATA, like GS 486 but far away better. we import directly from the producer at great prices so if anybody interested, please contact off-list. additionaly, we know another excellent producer (price of about 45$ FOB china) we would recomend off-list. The documentation and configuration screens look frighteningly like the Grandstream ones. Is this in fact the same hardware? If it has better firmware and software it might be a winner... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 alternative?
Ah! Now this is actually something I've not been able to get my head around: > Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which > has its own MP3 player. Can anybody tell me where this built-in MP3 player is in 1.2.x/how do I use it ? I still seem to have the usual two mpg123 processes running with 1.2.4, with whatever music on hold is set in musiconhold.conf I'm sure it is very obvious, but I can't for the life of me figure out what I'm supposed to do to use the built-in MP3 player facilities. I just have the following in my musiconhold.conf: [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 random=yes Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK X100P installation help
I just moved the card to another slot, and I now seem to have voice! Upon removing one of the cards, I noticed a resistor is broken off at the base. Not sure if it was always like that or if I did it removing it, but I now do seem to be further along. Still seems to be trouble detecting that the remote caller has hung up though. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul J. Smith Sent: 23 February 2006 17:18 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] UK X100P installation help Hi, I've got the latest [EMAIL PROTECTED] setup running with SIP. I've been trying to tie it in with the PSTN off and on for a while with no success. I gave up on ISDN and purchased 2 x100p cards from x100p.com. I've got the card installed, the machine can see it. The problems I have at the moment are First time I start up the drivers and asterisk, I can call the pstn number. The call is detected, the SIP phone rings, you can pick up the call, but there is nothing but a faint noise. After that, if you try to call the number again, it's engaged, as if the x100 P has not dropped the line, though watching the asterisk log in real time it says it has. I've looked all over the web for some detailed instructions, but most just mirror what I've already done. Is it because I'm in the UK? I could not find anyone using these in the UK. Maybe they don't like UK phone sockets? My zaptel.conf is # Span 1: WCFXO/0 "Wildcard X100P Board 1" fxsks=1 # Global data loadzone= uk defaultzone = uk Can anyone help me to get this working? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel CiscoHDLC / Fedore FC4
I am attempting to deploy a TE210P with one port configured (all 24 channels) as Cisco HDLC, currently running 2.6.11 FC4 with HDLC module compiled.HDLC (RAW) appears to work fine, however, using Cisco HDLC (via sethdlc & sethdlc-new) results in no traffic being encapsulated or transmitted via the hdlc0 interface. Is there anybody here currently running a similar configuration on a 2.6.x kernel or FC4? If so, could you let me know what kernel you're running, or what updates were required. I've done the standard make linux26 and necessary uncomments in zconfig.h etc, to no avail.Thanks,Anthony-- Anthony D Cennami ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem
Thanks for that Peter!
I think your message solved my problem: I set the master number to be in
group 1 (group=1) in capi.conf and called Dial with CAPI/g1 and it
worked perfectly.
However, with group=1 in capi.conf for the master number, at the moment
no matter what I do I'm getting the master number presented as the CLI.
This is fine by me because it is exactly what I want, but it is all very
confusing :-)
Faris.
Peter Braidwood wrote:
I have ISDN2 and 10 MSN numbers and also have just upgraded to 1.2.4 and
chan_capi-cm and have this working completely perfectly
Capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=en
[ISDN1]
isdnmode=msn
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=from-isdn
group=1
devices=2
bit of extensions.conf, I dial 9 for an outside line
[pstn]
exten => _9./321,1,Dial(CAPI/g1/01234567890:${EXTEN:1})
exten => _9./322,1,Dial(CAPI/g1/01234567891:${EXTEN:1})
exten => _9./323,1,Dial(CAPI/g1/01234567892:${EXTEN:1})
exten => _9./324,1,Dial(CAPI/g1/01234567893:${EXTEN:1})
exten => _9./326,1,Dial(CAPI/g1/01234567894:${EXTEN:1})
exten => _9./327,1,Dial(CAPI/g1/01234567895:${EXTEN:1})
exten => _9./328,1,Dial(CAPI/g1/01234567896:${EXTEN:1})
exten => _9./350,1,Dial(CAPI/g1/01234567897:${EXTEN:1})
exten => _9./351,1,Dial(CAPI/g1/01234567898:${EXTEN:1})
exten => _9./352,1,Dial(CAPI/g1/01234567899:${EXTEN:1})
So when extension 326 dials out the cli that is presented would be
01234567894
Contact me off list if you want any further help.
Peter Braidwood
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 23 February 2006 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem
I've having a big problem after having upgraded to 1.2.4 and
chan_capi-cm 0.6.4
When making outgoing calls I don't seem to have any control over the CLI
that is presented to the called party -- it can be any one of the MSNs
allocated to the line, allocated on what seems to be a random basis.
This is on a BT Business Highway line (which is essentially an ISDN2e
line with two built-in analog ports), configured with 8MSNs alongside
the single the "master" digital telephone number for the line.
With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk
1.0.9 it was always the "master" number that was presented, and that is
actually what I want.
Obviously the format of capi.conf has changed between these two versions
of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions
would be appreciated.
Here's my capi.conf (actual numbers changed to protect the innocent!)
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law
[123456]
; Master number for line - i.e. number for line before MSNs were
allocated
; and which still works and still accepts incoming calls.
isdnmode=msn
msn=01234123456
;incomingmsn=*
incomingmsn=123456
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
[123457]
;first MSN
msn=01234123457
;incomingmsn=*
incomingmsn=123457
isdnmode=msn
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
{repeated for next 7 MSNs}
And in extensions.conf I have:
[globals]
ISDN1=CAPI/123456
[mysip]
;GET OUTSIDE LINE (ISDN1 - dial 9)
ignorepat => 9
exten => exten => _9.,1,Dial(${ISDN1}/${EXTEN:1}/b)
exten => _9.,2,Playback(busy)
exten => _9.,3,Hangup
*
I've tried using ISDN1=CAPI/contr1 but it makes no difference.
I've tried leaving out the isdnmode=msn but it makes no difference
I've tried entering 01234123456 as the msn= line on all of the msn
entries in capi.conf but it makes no difference either.
And now I'm out of ideas and any help would be appreciated.
Thanks,
Faris.
p.s. sorry if this message is HTML. I've switched to using Thunderbird
and it is confusing the heck out of me. It claims this is a plain text
message but it doesn't look like plain text to me from this end!
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Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
On Feb 23, 2006, at 4:58 AM, Adam Robins wrote: Thanks, We already have a cron reboot of all of our Asterisk servers every night. We've been doing this for over a year due to memory leak issues. ??? What do you think this is windows 95??? I had a problem like that I would be looking at getting rid of asterisk. I don't ;~) I wonder what your leak is ? After 2 weeks of messing around with every conceivable IAX2 and jitterbuffer configuration, I switched to SIP yesterday. Complaints went from 10-20 per day to ZERO. Literally overnight. I wonder if this is an ILBC frame size issue of some sort? Seems odd. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
