Re: [asterisk-users] Question about SIP registration
I reply to your question below 1) I don't have a secret for that peer. 2) Obviously, the solution is to make the 'host' field static (in my scenario, because the port is non-standard 5080, so no standard endpoint SIP can register with that IPaddress:port) or specify a secret with 'host=dynamic'. The question I made was a little different: I'm wondering why an external SIP endpoint, which is trying to register on eth0 85.X.Y.Z network, is indeed seen by Asterisk as registered with address 1.1.1.1 (the eth1 IP addresses of the PC). I try to explain better: usually, SIP endpoint with IP address X.Y.Z.T which has registered itself on Asterisk (for example with user 200) is seen as following (sip show peers) 200/200 X.Y.Z.T5060OK(xx ms) So the CLI shows the *endpoint's* IP address. Instead, in my scenario, I see a row like this: 999/999 1.1.1.15060UNREACHABLE (1) And 1.1.1.1 is the eth1 IP address of the PC where Asterisk is installed on. But I haven't any endpoint SIP onto that PC which is trying to register, while I can see one of them OUTSIDE my network (i.e. in the Big Internet) that is trying to register as 999: in fact, if in [999] SIP account I put 'host=1.1.1.1', I can see a row like this on Asterisk log: [Jan 13 11:10:54] ERROR[1834]: chan_sip.c:8718 register_verify: Peer '999' is trying to register, but not configured as host=dynamic [Jan 13 11:10:54] NOTICE[1834]: chan_sip.c:15236 handle_request_register: Registration from '999 sip:9...@85.x.y.z ' failed for '174.129.74.46' - Peer is not supposed to register - while if I put 'host=dynamic' I saw (in sip show peers) the row depicted in (1) and no more errors like above. I suspect there is something wrong with network configuration (firewall, NAT). But this behavior is quite odd to me ... Alberto. PS: the network is at customer's site, so I haven't chance to have a clear look over it... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Lister Sent: martedì 12 gennaio 2010 18.51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about SIP registration On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote: Then I have configured an account as following: [999] type=friend username=999 You don't appear to have a secret= line in there with a password option... or did you snip it? Can someone explain me this kind of behaviour? Is it normal? Can I restrict registration of 999 peer only to SIP UA from network 1.1.1.X? There is an ACL option for the SIP peer which you can add, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip +permit-deny-mask (although there were some issues with this in earlier versions of asterisk.. it should work properly in recent versions.) Rob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about SIP registration
Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATted over public network, with address 89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) Then I have configured an account as following: [999] type=friend username=999 host=dynamic port=5080 context=sipfrom nat=no canreinvite=no call-limit=8 videosupport=no disallow=all allow=alaw qualify=15000 So far, so good. Now, I have an internal process (onto Linux PC) which is a SIP endpoint and should register to Asterisk as 1.1.1.1:5080, but an external entity (i.e. a SIP endpoint over public Internet) is trying to register to Asterisk as 9...@89.x.y.zmailto:9...@89.x.y.z:5060 and the registration SUCCEEDS! When I launch the CLI command sip show peers, I see a row like this: 999/9991.1.1.1 5060 OK (3 ms) Can someone explain me this kind of behaviour? Is it normal? Can I restrict registration of 999 peer only to SIP UA from network 1.1.1.X? Thanks for your help! Regards, Alberto Aggio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE
Hi, I have occasionally experienced the same problem too, and I suspect it was caused by some spikes in network traffic (e.g. for an intensive file transfer) that delayed too much SIP OPTION response, so that Asterisk marked these devices as UNREACHABLE; I was able to use the devices too: in fact, the only drawback is that other devices are not able to call the UNREACHABLE devices using Asterisk. The only solution I found was to disable 'qualify' field in SIP account, in order to put these devices in unmonitored state. Maybe it's not your problem, but you can monitor the network with a sniffer (e.g. ethereal), in conjunction with SIP debug in Asterisk (sip set debug) in order to check the correct arrival of OPTION response. Noevertheless, I'm wondering if there is another cause to this issue that is not depending on network, but on Asterisk itself, so let me know. HTH, cheers Alberto. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Sent: lunedì 4 gennaio 2010 22.13 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE Hi guys, Am having a strange SIP problem in my call centre. The call centre has about 70 SIP agents (some of the are using SIP hard phones, other SIP softphones), and occasionally most of the SIP peers (hardphones and softphones) become UNREACHABLE and then after few second again REACHABLE. Some hardphones and softphones work perfectly normal during that period (even normally responding to OPTIONS message), but most of them get UNREACHABLE. I don't have NAT - phones and Asterisk are in the same subnet, so nothing complicated really (regarding network configuration). I'm currently suspecting my network to be the problem, but I would just like to confirm with you guys, if you have any similar experiences, what could be causing this? Please, see bellow one of the sample SIP traces. Regards, Alex Jan 1 11:17:42 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060: OPTIONS sip:testpho...@165.11.1.41 SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport From: asterisk sip:aster...@165.11.1.50;tag=as02e1afaa To: sip:testpho...@165.11.1.41 Contact: sip:aster...@165.11.1.50 Call-ID: 3fa169320586bad01cd93bd87adf1...@165.11.1.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:45 VERBOSE[6046] logger.c: Retransmitting #1 (no NAT) to 165.11.1.41:5060: OPTIONS sip:testpho...@165.11.1.41 SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport From: asterisk sip:aster...@165.11.1.50;tag=as02e1afaa To: sip:testpho...@165.11.1.41 Contact: sip:aster...@165.11.1.50 Call-ID: 3fa169320586bad01cd93bd87adf1...@165.11.1.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:46 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now UNREACHABLE! Last qualify: 14 Jan 1 11:17:56 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060: OPTIONS sip:testpho...@165.11.1.41 SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport From: asterisk sip:aster...@165.11.1.50;tag=as796f6356 To: sip:testpho...@165.11.1.41 Contact: sip:aster...@165.11.1.50 Call-ID: 3367c4dc6cbdd57d67b0c5b53d549...@165.11.1.50 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:56 VERBOSE[6046] logger.c: -- SIP read from 165.11.1.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport From: asterisk sip:aster...@165.11.1.50;tag=as796f6356 To: sip:testpho...@165.11.1.41;tag=5A4BF5F8-460290A9 CSeq: 102 OPTIONS Call-ID: 3367c4dc6cbdd57d67b0c5b53d549...@165.11.1.50 Contact: sip:testpho...@165.11.1.41 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.0.0047 Content-Length: 0 Jan 1 11:17:56 VERBOSE[6046] logger.c: --- (10 headers 0 lines) --- Jan 1 11:17:56 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now REACHABLE! (16ms / 1ms) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
Re: [asterisk-users] Connect two Asterisk Server in IAX ?
Hi,
maybe this link can be useful:
http://www.voip-info.org/wiki/view/IAX+encryption
In particular, in your configuration I can't see the authentication method,
which must be md5, and a username to authenticate with, in either server.
But have a further look at the article, maybe you'll be able to sort out the
issue from that :)
HTH
//Al.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phibee Network
Operation Center
Sent: sabato 21 novembre 2009 8.16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Connect two Asterisk Server in IAX ?
Hi
My first post get no answer :=, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
calltokenoptional=192.168.0.20
[Srv2]
type=peer
host=192.168.0.20
qualify=yes
trunk=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
context=Incoming
peercontext=Incoming
extension.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
[Incoming]
exten = _X.,1,Playback(demo-thanks)
exten = _X.,2,Hangup
[Out]
exten = _201X.,1,Dial(IAX2/Srv2/${EXTEN:3},90,r)
exten = _201X.,2,Congestion
==
Srv1*CLI iax2 show peers
Name/UsernameHost Mask Port Status
Srv2 192.168.0.20 (S) 255.255.255.255 4569 (E) OK (39 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
On Srv2
iax.conf
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
calltokenoptional=192.168.0.5
bandwidth=low
[Srv1]
type=peer
host=192.168.0.5
qualify=yes
trunk=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
context=Incoming
peercontect=Incoming
extensions.conf:
[Incoming]
exten = _X.,1,Playback(demo-thanks)
exten = _X.,2,Hangup
[Out]
exten = _202X.,1,Dial(IAX2/Srv1/${EXTEN:3},90,r)
exten = _202X.,2,Congestion
===
trader-voip*CLI iax2 show peers
Name/UsernameHost Mask Port Status
Srv1 192.168.0.5 (S) 255.255.255.255 4569 (E) OK (28 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
===
All SIP Poste are connected and have in context in: Out
Now, when i call from a post connected on Srv1, i have this error on Srv1:
[Nov 21 08:09:44] WARNING[6407]: chan_iax2.c:9018 socket_process: Call
rejected by 192.168.0.20: No authority found
and on Srv2:
[Nov 21 08:09:44] NOTICE[9089]: chan_iax2.c:9785 socket_process:
Rejected connect attempt from 192.168.0.5, who was trying to reach
'1...@incoming'
125 are the number called (201125)
Dialplan on Srv2
Srv2*CLI dialplan show Incoming
[ Context 'Incoming' created by 'pbx_config' ]
'_X.' = 1. Playback(demo-thanks)
[pbx_config]
2. Hangup()
[pbx_config]
-= 1 extension (2 priorities) in 1 context. =-
Anyone can help me for know where is my error ?
thanks
Jerome
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Re: [asterisk-users] destroy zombie session
Sorry for late response. I had to reproduce problem (it's not systematic). When iax2 show channels gives this output IP-AM-PBX*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None)10.229.47.113REMOTE_SER 06818/14174 2/2 0ms -0001ms ms unknow 1 active IAX channel core show channels gives the following output: IP-AM-PBX*CLI core show channels Channel Location State Application(Data) 0 active channels 0 active calls So I can't gather any information from the last command ... any ideas? Thanks, //Al. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: venerdì 13 novembre 2009 17.38 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] destroy zombie session What does the zombie call look like in core show channels? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aggio Alberto Sent: Friday, November 13, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] destroy zombie session Hi all, Some time ago I posted an issue regarding the hangup of active calls from the CLI and someone told me that soft hangup should work. Well, in fact it does work, but only if the channel is known, i.e. it doesn't work for zombie channels. For example, I have this scenario (CLI output of command iax2 show channels) IP-AM-PBX*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None)10.229.47.113REMOTE_SER 06818/14174 2/2 0ms -0001ms ms unknow 1 active IAX channel IP-AM-PBX*CLI Here I can't issue soft hangup command because I haven't a channel to specify (None is not a choice :) ). Now the question is: is there a way to drop this (zombie) channel off and release frozen resources? Restarting asterisk is not an option (or maybe the last chance if I have no other way to achieve this result :)) Thanks in advance for your replies. Cheers, Alberto Aggio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP traffic through Asterisk??
As far as I could try some solutions, the only one that works as you like involved use of Transfer() application, defining as 'tecnology' something like that: SIP/exten@ip_address Where ip_address is the address of the peer you want to transfer the call to. By the way, I found a scenario where this trick still keeps not working: if the transferor (i.e. the caller) is a registered SIP user, I saw that the transfer is done, but Asterisk is still in the path. Vice versa, if the caller is NOT a registered user, the transfer will exclude asterisk from the path either if the transferree (i.e. third party called) is registered to Asterisk or not. HTH Alberto. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ignacio Sent: martedì 17 novembre 2009 14.07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic through Asterisk?? Thank you very much to both of you. My problem was that I used transfer in the dialplan. I have read that If I have Tt, wW, or hH, then asterisk will always stay in the path. So I have to redefine what I want to do know. Allowing transfers is an useful feature, but I wanted all rtp traffic went p2p. Is there any intermediate solution? Thanks. Regards Ignacio On Mon, Nov 16, 2009 at 7:52 AM, Leonja Cerebro lio...@gmail.com wrote: see the DTMF method on both phones. 2009/11/14 Ignacio sanfermi...@gmail.com Ok, thank you very much. I will try to find any information in asterisk documentation about RTP. On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote: I have just established a call between 2 sip phones and I have noticed that all RTP traffic goes through Asterisk Server. I was expecting RTP traffic went to one phone to another phone directly. I set canreinvite=yes in sip.conf in both sip peers. I also tested it with 2 mgcp phones and same result, all rtp traffic goes through Asterisk. Is there any way to force traffic to go from one phone to another? snip I don't recall where it is off-hand but, somewhere in the Asterisk documentation, there is an explanation of how Asterisk makes a decision about reinvites. You may want to look at that to see if your environment satisfies all the requirements and how it can be adapted if it does not - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- We never did too much talking anyway So don't think twice, it's all right -- There are more things in heaven and earth, Horatio, Than are dreamt of in your philosophy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] destroy zombie session
Hi all, Some time ago I posted an issue regarding the hangup of active calls from the CLI and someone told me that soft hangup should work. Well, in fact it does work, but only if the channel is known, i.e. it doesn't work for zombie channels. For example, I have this scenario (CLI output of command iax2 show channels) IP-AM-PBX*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None)10.229.47.113REMOTE_SER 06818/14174 2/2 0ms -0001ms ms unknow 1 active IAX channel IP-AM-PBX*CLI Here I can't issue soft hangup command because I haven't a channel to specify (None is not a choice :) ). Now the question is: is there a way to drop this (zombie) channel off and release frozen resources? Restarting asterisk is not an option (or maybe the last chance if I have no other way to achieve this result :)) Thanks in advance for your replies. Cheers, Alberto Aggio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pattern matching DID
Hi, it's quite straightforward: you can do your dialplan like this (default is the default context answered when inbound calls happen) - remember the underscores! - [default] exten = _1703,1,Goto(place-IVR,s,1) exten = _1567 ,1,Goto(place-other,s,1) [place-IVR] exten = s,1,Answer exten = s,2,Background(menu-file) exten = 1,1,Goto(submenu,1) exten = 2,1,Goto(submenu,2) (...) [place-other] exten = s,1,Answer exten = s,n,... (...) exten = s,n,Hangup If you want to jump into a specific part of context, you should put a label near the 'n' priority where you want to jump to (eg. exten = s,n(jumphere),application/function) then specify that label into Goto() application. Cheers, //Al. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: domenica 1 novembre 2009 21.46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] pattern matching DID I have two DID numbers. I want callers who dial 1 703 to get placed in a specific part of IVR I want other callers who dial 1 567 to get placed in a different area. How do I do this please? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Clear pending SIP channels
Hi all, I have a question regarding pending (zombie) SIP sessions: on Asterisk CLI, with command 'sip show channels' , I see two channels in use with callID and other infos detailed; also 'sip show inuse' give me same result (in terms of channels usage): PeerUser/ANRCall ID Seq (Tx/Rx) Format Hold Last Message xx.xx.xx.79 209 1745914a212 00102/0 0x8 (alaw) No Tx: ACK xx.xx.xx.34 217 3c515bbb7c8 00101/2 0x8 (alaw) No Rx: ACK * Peer name In use Limit 997 0/0 2 (...) 217 1/0 2 216 0/0 2 215 0/0 2 214 0/0 2 213 0/0 2 212 0/0 2 211 0/0 2 210 0/0 2 209 1/0 2 208 0/0 2 (...) 200 0/0 2 is there a way to drop these calls throughout the CLI or I have to restart asterisk? Many thanks in advance and regards, Alberto Aggio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call retard from a softphone to a hardphone
Hi group I'm newbie on Asterisk so I followed the Linux Networking CookBook by Carla Schroder to make my first call. My asterisk box is on a Debian box with an public static IP. The clients (2) are with dynamic private IP's I'm using SJphone on a PC and a Linksys PAP2-NA to make calls between them. Both of them register well on my Asterisk server but when I call from the SJPhone to the PAP2 the voice comes with retard, and progressively the voice is bad. This is my sip.conf [general] context=default port=5060 bindaddr=0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw ;SARAHC is the PAP2 [sarahc] ;Sarah Connor type=friend username=sarahc secret=5656 host=dynamic disallow=all allow=alaw allow=ulaw dtmfmode=rfc2833 outgoinglimit=1 context=local-users ;DUTCHS is the SJPhone [dutchs] ;Dutch Schaeffer type=friend username=dutchs secret=6767 host=dynamic context=local-users Sorry if I'm not giving enough information because I'm new to this wonderful tool but any idea or guide would be very good. Thanks in advanced Carlos Bernat ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Packetcable
Hi group I wrote 2 years ago to know if there is some workaround for PacketCable. Since then I got no answer and now I hope there's something about. Is there any chance to use Asterisk as softphone with cable modem technology using Packetcable? Thanks in advanced Carlos Bernat ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information
Sean Dennis ha scritto: Sigma Networks wrote: ... My current questions are: 1. How to remotely reboot 7970s. I have both web access and SSH access to the phones. The instructions I have for SSH are to use (1) user/pass (or whatever is in the confg) and then (2) debug/debug. Surprisingly reset is not a valid command to restart the phone. There doesn't appear to be a reset on the web page, maybe there's a hidden URL? 2. BusyLampField? ... We have about 200 79x1's running SIP w/ asterisk and we are very pleased despite some of the non-standard things Cisco does. In answer to question 1 the only way we have found to reboot the phone remotely is shutdown the port on the POE switch. This will drop the PC's network as well if it is plugged into the phone. Question 2 I would like to know the answer to myself. I would be curious to know if it works with the SIP image in call manager. Same here. We have about 500 phones, from both 79x1 and 79x0 series; I posted the same two questions twice some time ago but never got an answer: I do reboot phones by power cycling them too, while I've been able to use blf with sccp images only. Furthermore, XML Services on 7940/7960 seem to be broken or at least to behave in different way than the one described in the sdk documentation. I needed the reboot feature to implement extension mobility but I wasn't able to find a clean way. Power cycling is not always an usable method, as many phones are powered by the AC adaptor. I think I will able to put my hands on an UCM6.1 box very soon to try that out and eventually grab the xml profiles. As soon as I get the info I'll surely post it on this ML and on voip-info too. Alberto. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards
Ron Joffe ha scritto: On Friday 01 February 2008 15:31, Matt wrote: It's about time Digium got on the ball and made PCI-e cards. What are people's experiences with this card? Anyone know if there are plans for a PCI-e analog card for FXO use? I have been using 220B's for about 6 months. I have about 20 of them out in the field. I have not had any issues with them, and feedback is positive. Same here. I've been using five TE220B in my company at 5 different sites since october 2007; up to now, zero problems and no echo at all. One of the sites runs a small callcenter that handles about 1000 incoming calls per day. So far the feedback is really positive. Alberto. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA with pulse dialing support over FXS
Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards from the plastic case and to fit them into the phones after making the appropriate changes to the phones' exterior to add holes for rj-45 socks and dc power input) Thanks. -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI card with PCI-E interface
Olivier ha scritto: Hi, Does such card exist ? It seems all existing models are designed for PCI buses. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The Sangoma A500BRX is a 2 to 6 bri pci-x interface, although I've never tested it (A500BRECX comes with hw echo cancellation). Regards, Alberto. -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] autoprovision 200+ linksys phones setup
Erick, I'm not aware of any precompiled package with such capabilities. However, I've implemented this capability myself by creating a python script that generates automatically tftp profiles for linksys phones (941, 942, 922, etc.), as well as entries for sip.conf, extensions.conf, voicemail.conf from a csv text files containing various fields such as mac address, display name, extension (sip user), sip password, etc. I've got some support from linksys themselves by registering as a Linksys installer for Italy. They have an autoprovisioning guide (PDF) with a command line compiler for encrypting tftp profiles. This compiler has a very useful command line option --sample-xml-compact; it creates a sample xml profile with all parameters supported by a given firmware release. (I don't use encrypted profiles, so the compiler is really only a starting point for creating profile templates) I have the win32 version for firmware 5.1.15 (spa942-5-1-15-spc-win32-i386.exe) Below is the sample xml generated by this tool. If you need additional info (such as the pdf guide or other stuff) email me privately, as I can't post this on the mailing list Bye, Alberto. Sample XML profile for linksys spa-942/922, firmware release 5.1.15 (some text lines got wrapped, if you cut paste them, remove the annoying linefeed) ?xml version=1.0 encoding=UTF-8 standalone=yes? flat-profile xmlns=http://www.sipura.net/xsd/SPA942; xmlns:xsi=http://www.w3.org/2001/XMLSchema-instance; xsi:schemaLocation=http://www.sipura.net/xsd/SPA942 http://www.sipura.net/xsd/SPA942/SPA942-5-1-15.xsd; Restricted_Access_Domains ua=na/Restricted_Access_Domains Enable_Web_Server ua=naYes/Enable_Web_Server Web_Server_Port ua=na80/Web_Server_Port Enable_Web_Admin_Access ua=naYes/Enable_Web_Admin_Access Admin_Passwd ua=na/Admin_Passwd User_Password ua=rw/User_Password Connection_Type ua=rwDHCP/Connection_Type Static_IP ua=rw/Static_IP NetMask ua=rw/NetMask Gateway ua=rw/Gateway PPPoE_Login_Name ua=rw/PPPoE_Login_Name PPPoE_Login_Password ua=rw/PPPoE_Login_Password PPPoE_Service_Name ua=rw/PPPoE_Service_Name HostName ua=rw/HostName Domain ua=rw/Domain Primary_DNS ua=rw/Primary_DNS Secondary_DNS ua=rw/Secondary_DNS DNS_Server_Order ua=naManual/DNS_Server_Order DNS_Query_Mode ua=naParallel/DNS_Query_Mode Syslog_Server ua=na/Syslog_Server Debug_Server ua=na/Debug_Server Debug_Level ua=na0/Debug_Level Primary_NTP_Server ua=na/Primary_NTP_Server Secondary_NTP_Server ua=na/Secondary_NTP_Server Enable_VLAN ua=rwNo/Enable_VLAN VLAN_ID ua=rw1/VLAN_ID Enable_CDP ua=naYes/Enable_CDP Provision_Enable ua=naYes/Provision_Enable Resync_On_Reset ua=naYes/Resync_On_Reset Resync_Random_Delay ua=na2/Resync_Random_Delay Resync_Periodic ua=na3600/Resync_Periodic Resync_Error_Retry_Delay ua=na3600/Resync_Error_Retry_Delay Forced_Resync_Delay ua=na14400/Forced_Resync_Delay Resync_From_SIP ua=naYes/Resync_From_SIP Resync_After_Upgrade_Attempt ua=naYes/Resync_After_Upgrade_Attempt Resync_Trigger_1 ua=na/Resync_Trigger_1 Resync_Trigger_2 ua=na/Resync_Trigger_2 Resync_Fails_On_FNF ua=naYes/Resync_Fails_On_FNF Profile_Rule ua=na/spa$PSN.cfg/Profile_Rule Profile_Rule_B ua=na/Profile_Rule_B Profile_Rule_C ua=na/Profile_Rule_C Profile_Rule_D ua=na/Profile_Rule_D Log_Resync_Request_Msg ua=na$PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH/Log_Resync_Request_Msg Log_Resync_Success_Msg ua=na$PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH/Log_Resync_Success_Msg Log_Resync_Failure_Msg ua=na$PN $MAC -- Resync failed: $ERR/Log_Resync_Failure_Msg Report_Rule ua=na/Report_Rule Upgrade_Enable ua=naYes/Upgrade_Enable Upgrade_Error_Retry_Delay ua=na3600/Upgrade_Error_Retry_Delay Downgrade_Rev_Limit ua=na/Downgrade_Rev_Limit Upgrade_Rule ua=na/Upgrade_Rule Log_Upgrade_Request_Msg ua=na$PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH/Log_Upgrade_Request_Msg Log_Upgrade_Success_Msg ua=na$PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR/Log_Upgrade_Success_Msg Log_Upgrade_Failure_Msg ua=na$PN $MAC -- Upgrade failed: $ERR/Log_Upgrade_Failure_Msg License_Keys ua=na/License_Keys GPP_A ua=na/GPP_A GPP_B ua=na/GPP_B GPP_C ua=na/GPP_C GPP_D ua=na/GPP_D GPP_E ua=na/GPP_E GPP_F ua=na/GPP_F GPP_G ua=na/GPP_G GPP_H ua=na/GPP_H GPP_I ua=na/GPP_I GPP_J ua=na/GPP_J GPP_K ua=na/GPP_K GPP_L ua=na/GPP_L GPP_M ua=na/GPP_M GPP_N ua=na/GPP_N GPP_O ua=na/GPP_O GPP_P ua=na/GPP_P GPP_SA ua=na/GPP_SA GPP_SB ua=na/GPP_SB GPP_SC ua=na/GPP_SC GPP_SD ua=na/GPP_SD Max_Forward ua=na70/Max_Forward Max_Redirection ua=na5/Max_Redirection Max_Auth ua=na2/Max_Auth SIP_User_Agent_Name ua=na$VERSION/SIP_User_Agent_Name SIP_Server_Name ua=na$VERSION/SIP_Server_Name SIP_Reg_User_Agent_Name ua=na/SIP_Reg_User_Agent_Name
[asterisk-users] Extension Mobility with Asterisk and Cisco 79x1 phones
Hi. I'm trying to develop a module that emulates the Cisco Extension Mobility feature from CallManager (the ability to log in to a phone and temporarily acquire the extension, soft key programming, and all other settings for that user profile) with Asterisk 1.4 and Cisco 79xx phones (some with SIP and some with SCCP, as the 7914 extension module does not support SIP). I've almost completed all the pieces of the puzzle (using a mix of xml pages and python scripts) but I'm stuck at one point: how to tell a java based cisco phone (e.g. 7911, 7961, 7970) to reload immediately from tftp its profile without rebooting? I know that the check-sync event in a sip notify message would do that, but this method has some disadvantages: - it is triggered after about 20 seconds the sip notify message has been received - the reboot is complete and it takes a lot of time I know for instance that on AAstra 5xi phones it is possible to push this settings on the fly without rebooting the phones. Can the same be done with Cisco Unified IP Phones? Is there a URL inside the phone to which I can make a GET or POST request from asterisk? Thanks, Alberto. -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Modem bridging on Asterisk (no VoIP involved)
Hi everybody. I know maybe this question has been posted some time ago, but I need your updated opinion on the subject. I'm replacing our old pbx with asterisk. I have two TE207 dual pri (e1) cards on a clustered system (one on each node). I absolutely need to connect 4/5 analog extensions with modems, they're being used for remote assistance on very old systems which cannot be upgraded to native IP links. Is there a good hardware that can bridge the e1 lines on the digium te207 card to my modems? A PCI card? An external box? I don't want to relay modem connections over ip, I just need to bridge them internally on the asterisk server: E1 == TE207 == Asterisk == (some hardware with FXS) == modems TIA for your replies. -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7970 CTLFile.tlv?
Matthew Rubenstein wrote: I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Cisco firmware (both sip and sccp) got me several headaches. When the phone finds something it doesn't like in the xml profile, (even a stupid thing such as a string too long), 99% of the times will abort the configuration download process and reboot in the worst case, or keep the previous one (if any). XmlDefault.cnf.xml gets loaded ONLY if the phone cannot find the relevan SEPMAC.cnf.xml. I think that renaming the SEP file to XmlDefault doesn't solve the problem. You should have a confirmation by checking out the Status/Messages menu on the phone (Settings button, then 4, 1) You have to find what's wrong inside your xml file. Don't worry about tlv files, they're optional and required only if you need to upload certificate trust lists. My suggestion is to cut down the SEP file to its simplest form (e.g. containing only tags with ssh username and password, and firmware version) in order to let you log on to the phone (even if it's not registered) via ssh and use the debug/debug, log/log, or user/default post-ssh logins; that should (maybe) let you get a hint from the phone's internal log. Just for example I noticed that on my 8.3.0 sip firmware, although specs tell that phone label is 12 chars long, if I enter more than 11 characters in the xml tag, the phone won't load the configuration (of course no mention for that fact anywhere). Alberto. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup
Benny Amorsen ha scritto: O == Olivier [EMAIL PROTECTED] writes: O 2. From this list, the WiFi hardphones which got only positive O answers where Siemens Gigaset SL75 and Nokia EXX Series. The Nokia SIP client isn't particularly impressive. However, most of its problems can be solved by telling the phone to use Asterisk as a proxy and not fill out the register settings (except server, which should just be sip:asterisk). This is obviously wrong, as Asterisk isn't a SIP proxy, but it works. NAT is a problem; you cannot specify STUN servers, but the phone will try to find them based on SRV records on the server address you specify. It will also try the default STUN port on the SIP server. /Benny I also found the Pirelli DP-L10 dual phone to be an excellent sip client with good roaming support and discrete battery saving capability. (Used in a 14-cell wifi network with 40 cellphones). Alberto. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bridge two connected calls
Nick Seraphin wrote: ... Once the incoming caller is in the dialplan, issue a Dial() command using both the m option and the M() option, in addition to any other options you would normally be using for Dial(). The m option will play music on hold while the Dial() command does it's thing. ... It works like a charm. Thanks a lot for the precious hint. Alberto. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to bridge two connected calls
Hi everybody. I am in the following scenario: 1 Customer A calls an asterisk box over a Zap channel on a toll free number during night time 2 The incoming call enters an AGI script on the dialplan 3 The AGI script plays back a welcome message, then starts the music-on-hold stream 4 The AGI script originates a calls to a stand-by operator's cell phone (operator B) 5 When the operator B answers the call, he is prompted (via another AGI script in the dialplan) to dial 1 to be recognized as human (the AMD() function is too random to be useful) 6 After being recognized as human, Customer A must be bridged to Operator B Everything is ok from 1 to 5, but I cannot really figure out how to accomplish task #6 I've tried with MeetMe or call parking but with no success. Can anyone point me in the right direction? Thanks -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Announcement file is unavailable?????
Folks, please, take a look at this asterisk log message: [Oct 2 08:55:13] WARNING[10290] app_queue.c: Announcement file 'atcert' is unavailable, continuing anyway... [Oct 2 08:55:13] WARNING[10290] app_queue.c: Agent on Agent/1001002 hungup on the customer. but: -bash-3.1$ whoami asterisk -bash-3.1$ ls -ls $HOME/sounds/atcert* 12 -rw-r--r-- 1 asterisk asterisk 10956 Oct 2 07:00 /var/lib/asterisk/sounds/atcert.gsm -- agent_call, call to agent '1001019' call on 'IAX2/va001019-4' -- IAX2/va001019-4 Playing 'beep' (language 'en') -- Agent/1001019 answered Local/[EMAIL PROTECTED],2 -- Agent/1001019 Playing 'atcert' (language 'en') -- Stopped music on hold on Local/[EMAIL PROTECTED],2 In other words, the file exists but it is played in a irregular basis, why? Could be an application bug? Because of this we are having a huge number of unanswered calls. Regards. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA400 and asterisk
You could use it as a usually FXO Gateway. I have tested and it works fine. 2007/6/12, MBIT Technologies [EMAIL PROTECTED]: Hi Guys I am just looking to see if you can help me. I have been investigating the SPA400 and it seems to run asterisk for the voicemail system. Does anyone know if it could be programmed to also talk to the FXO ports? ___ --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 -- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --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] WiFi SIP phones
You could try, N80, N95 devices. It cost arround 300 dollars and works fine with SIP , Wifi and GSM. I have been trying for several weeks with Truphone, Gizmo, Asterisk and other providers my N80 IE, and it works perfectly Regarsd 2007/5/23, Chris Bagnall [EMAIL PROTECTED]: Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since then. What models are currently out there people would recommend I look at? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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 -- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --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] WiFi SIP phones
After trying painfully many many many flawed devices we eventually found a very good solution for our wi-fi network: Pirelli DP-L10. It's a dual GSM/Wi-Fi SIP triband (800/1800/1900). As a cellphone, no special note about it, but as a wifi phone, it has two main features which make it worth using: - excellent battery life - excellent roaming support between different APs even with no L2 fast roaming support or WPA key caching We now have 30 phones registered on an asterisk 1.2 server, connected through a 14 access point network (using wpa-psk). We've reached a DECT-level quality! Alberto Sagredo (M) ha scritto: You could try, N80, N95 devices. It cost arround 300 dollars and works fine with SIP , Wifi and GSM. I have been trying for several weeks with Truphone, Gizmo, Asterisk and other providers my N80 IE, and it works perfectly Regarsd 2007/5/23, Chris Bagnall [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since then. What models are currently out there people would recommend I look at? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] test
ACK 2007/4/12, Razza [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 -- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --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] Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
send channel to server 10.0.10.5:5060, handle = 8 local port= 0 sipTransportSendMessage:Sent SIP message to 10.0.10.5:5060, handle=8, length=360, message= ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5 From: Cisco 7940 sip:[EMAIL PROTECTED];tag=0013c3677fdf00ae6752cb07-7fbc304d To: sip:[EMAIL PROTECTED];user=phone;tag=as1ae4df20 Call-ID: [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 13:39:57 GMT CSeq: 101 ACK Content-Length: 0 sipTransportSendMessage: Closed a one-time UDP send channel handle = 8 Proxy-Authenticate= Digest algorithm=MD5, realm=msoft, nonce=733b51d0 sipSPISendInviteMidCall: Sending INVITE... sipSPIGenRequestURI: Forming Req-URI (Caller): using original Req-URI get_next_request_trx_index: Getting next TRX index, sent = 1 get_next_request_trx_index: Got TRX(0) for sent req get_last_request_trx_index: Getting last TRX index, sent = 1 get_last_request_trx_index: Got TRX(0) for sent req get_last_request_trx_index: Getting last TRX index, sent = 1 get_last_request_trx_index: Got TRX(0) for sent req SIPSPIAddRouteHeaders: Route info not available; will not add Route header. get_last_request_trx_index: Getting last TRX index, sent = 1 get_last_request_trx_index: Got TRX(0) for sent req sipTransportSendMessage: ccb 0: config 10.0.10.5:5060 - remote 10.0.10.5:5060 sipTransportSendMessage: Got handle 2 sipTransportSendMessage: Opened a one-time UDP send channel to server 10.0.10.5:5060, handle = 8 local port= 0 sipTransportSendMessage:Sent SIP message to 10.0.10.5:5060, handle=8, length=1224, message= INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK2e21f6c7 From: Cisco 7940 sip:[EMAIL PROTECTED];tag=0013c3677fdf00ae6752cb07-7fbc304d To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 Date: Thu, 12 Apr 2007 13:39:57 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7940G/8.0 Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Proxy-Authorization: Digest username=215,realm=msoft,uri=sip:[EMAIL PROTECTED];user=phone,response=25d8a11faab3a8e3ff6c7fa74f142475,nonce=733b51d0,algorithm=MD5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: Cisco 7940 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 19361 0 IN IP4 10.0.10.136 s=SIP Call t=0 0 m=audio 16946 RTP/AVP 8 0 18 101 c=IN IP4 10.0.10.136 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv sipTransportSendMessage: Closed a one-time UDP send channel handle = 8 LINE 0/1: sipTransportSendMessage: Stopping reTx timer LINE 0/1: sipTransportSendMessage: Starting reTx timer (500 msec) SIPTaskProcessListEvent: cmd = 0x0 sip_sm_process_event LINE 0/1: --0x000557d9-- : SIP_STATE_SENT_INVITE - E_SIP_TIMER sipTransportSendMessage: ccb 0: config 10.0.10.5:5060 - remote 10.0.10.5:5060 sipTransportSendMessage: Got handle 2 sipTransportSendMessage: Opened a one-time UDP send channel to server 10.0.10.5:5060, handle = 8 local port= 0 sipTransportSendMessage:Sent SIP message to 10.0.10.5:5060, handle=8, length=1224, message= ..many other retransmissions follow -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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 1.2.17 and Cisco 7940/SIP: bug or what?
Doug Lytle ha scritto: Alberto Pastore wrote: Firmware on 7940 is 8.6 (the latest one). I had the same issue. I ended up moving back to firmware P0S3-07-4-00 on the phone. I did a telnet into the phone, did a show register and shaw some very weird info. Normally, I would see: ... But why does 8.6 seem to work with previous asterisk 1.2.13?? ___ --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] Pound # key not being handled
I am trying to use call parking. I have the following in features.conf [general] parkext = 700 parkpos = 701-720 context = parkedcalls When I try #700 from my softphone asterisk just passes it and doesn't interpret it. Can someone tell me what I am missing? I am using asterisk-1.2.17 Thanks, Alberto ___ --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] moving WiFi phone
Pavel Jezek ha scritto: Jens Vagelpohl wrote: I have two APs (Apple AirPorts) sending on the _same_ channel. Handover works perfect with no discernible loss of connectivity or audio using a Siemens SL75. The handover cannot even be noticed. as I know, best practice says, that neighboring AP should use _non overlapping_ channels... :-\ After *months* of troubles using a 14 APs network with same SSID, WPA/TKIP security model, tx power settings and channels carefully distributed in order to be as non overlapping as possible, including a controller capable of performing fast layer2 reauthentication (e.g. something like caching WPA keys between access points), I always got VERY POOR roaming performance. I've tested these phones: UTStarcom F1000 UTStarcom F1000g UTStarcom F3000g Siemens Gigaset SL75 WLAN Nokia E60 Nokia E70 Samsung WIP6000 Linksys WIP300 I was desperate. I took a bold step. I downgraded to WEP-128 (I know it's weak) and, despite the recommendations from any good wifi networking guide, I SET ALL APs ON THE SAME CHANNEL. Don't ask me why, but now roaming is PERFECT, never had a call dropped or even a hiss or crackling noise during conversation. I can even run or move over the site hangar on forklift trucks while talking on the phone, at 15-20 mph. Luckily there are no high throughput demands for data transmission (PDAs, notebooks, etc) over the wifi network, so I didn't got performance issues. Imho roaming support on 802.11 wifi networks is far from being usable... -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] Cordless SIP Phones
Siemens Gigaset IP phones (C450-IP, S450-IP) are not that bad (gigaset.siemens.com). C450IP costs less than 100 USD (in Italy at least), S450 is slightly more expensive. Grandstream HT-286 works quite well with DECT handsets too. (I've deployed both and both are working with Asterisk). Alberto. [EMAIL PROTECTED] ha scritto: I share your frustration. Might I suggest a Grandstream HT-386 (or 486, etc) gateway to a regular cordless phone? On Sun, Jan 28, 2007 at 09:52:19PM -0500, Edward Halman wrote: Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you. Ed ___ --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] Thomson ST2030S and BLF
Andrew Joakimsen ha scritto: I know of the call pickup issues but what asterisk issue and what BLF issue? On 1/25/07, Alberto Pastore [EMAIL PROTECTED] wrote: Andrew Joakimsen ha scritto: Actually I noticed just three days ago there is a new release, and the releae notes seem to address Disable TrMail and Pickup keys Disable call progress indication ___ but it does not address poor guys' troubles with asterisk, blf and call pickup... If you configure a thomson key to supervise a line, with the proper hints in extensions.conf, BLF works great. Unfortunately, if you press, by mistake or choice, a flashing key (when the related sip extension is ringing): 1-you won't pickup the call, as it fails 2-the key will remain flashing and useless until you reboot the phone ___ --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] Thomson ST2030S and BLF
Andrew Joakimsen ha scritto: Actually I noticed just three days ago there is a new release, and the releae notes seem to address Disable TrMail and Pickup keys Disable call progress indication ___ but it does not address poor guys' troubles with asterisk, blf and call pickup... BTW, anyone of you having problems also with RTTTL melodies? My ST2030S phones seem to playback a RTTL melody at 1/5 its original speed. (I know this is not vital, but as soon as my users discover the possibility of uploading rtttl ringtones, they begin annoying me by asking how they work) ___ --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] About BRI / ISDN hardware. What to buy?
Jens Vagelpohl ha scritto: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 18 Jan 2007, at 18:31, Patrick wrote: I think http://www.melware.de carries the Eicon Server ISDN cards which have hardware echo cancellation. They are also the author of the chan_capi driver for Asterisk. I use the Eicon Server BRI cards with Asterisk myself and they work very well. I concur, I have a Eicon DIVA single port BRI card and it works very well. Cosmin, if you want to use it for Fax traffic as well make sure you do *not* get a V-BRI card. Those will not do Fax. jens Tried almost all cards (Junghanns, Sangoma, Beronet, some hfc-based oem cards, Eicon Diva Server). Eicon is expensive but is *REALLY* worth it. The other cards are just a waste of money (even if little money). If you want a reliable PBX (who doesn't want it?), Diva Server cards are the definitive choice. The best card ever. Zero echo problems, superb hardware echo cancellation. Top reliability. Excellent FAX support with Hylafax (only cards with builtin DSPs, that is, NOT the V-series, as pointed out by Jens). Easy driver installation and powerful utilities/configuration tools. I tested BRI-2M, 4BRI-8M, PRI-30M on several installations, even older 1.0 version cards (PCI 5v only) just work great. I use diva server drivers software source rpm from Eicon, chan_capi from www.melware.org (0.7.1) on asterisk 1.2.14 (kernel 2.6.17.3). We've deployed more than 40 PBX (from 1 bri to 8 bri) without a flaw. I'm only a little bit annoyed about not being able to take advantage of the onboard DSPs to perform audio transcoding, because of the lack of a suitable asterisk driver (the cards themselves support hardware gsm/g726 codecs, for instance). Alberto. ___ --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] About BRI / ISDN hardware. What to buy?
Cosmin Prund ha scritto: I do not care about fax. I just want a good VOICE card. Can someone please give a price quote for this card, give or take 10%? I just spent 5 minutes filling in a really long form on a shopping web site to get a price quote, only to find my account needs to be manually activated before I can see the price! That's *STUPID*. If I have a choice, I'll buy it from somewhere else... BTW, check out on eBay (search for diva server) you could make good bargains. Alberto. ___ --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] About BRI / ISDN hardware. What to buy?
On Gio, Gennaio 18, 2007 20:16, Armin Schindler wrote: Sorry for that ;-) chan-capi already has rtp code to select one codec using also DSPs anti-jitter buffer. It is not fully tested and the full support for all codecs is still missing. Also, full conferencing using the DSPs is in progress. Armin Wow! Well, maybe annoyed was the wrong term, as I don't really-really miss the transcoding feature. Let's just say it would be better to have it. Surely, I will test any test release I can put my hands on. Keep up with the good work! Thanks, Alberto. ___ --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] About BRI / ISDN hardware. What to buy?
On Gio, Gennaio 18, 2007 20:44, Armin Schindler wrote: On Thu, 18 Jan 2007, Cosmin Prund wrote: I finally found a price tag for the darn thing, at around 500 euros I can handle it. Qustion: Do they behave properly if I've got an other Digium TDM400 card in the system? How about installing two cards in the same server? At the moment I've only got 1 ISDN line plus a few analog lines going into the TDM but in the very near future we might want to get a second ISDN. The DIVA Server cards and its driver supports multiple cards. As long as other cards behave correctly on the bus, I don't see any problem here. Armin I'm not aware of any conflict between Digium TDM and Diva Server, although I did not test that configuration. Anyway, I've set up some asterisk boxes with 2/3 DIVA-2M, even one with a Diva BRI-2M 1.0 and a DIVA BRI-2M 2.0 in the same PC, as well as one server with two DIVA 4BRI-8M 1.0 in it. Whe you run Eicon Config utility, the software recognizes instantly all cards models, and numbers each port sequentially, so that you can access them in asterisk individually or as a group by properly configuring chan-capi via capi.conf and Dial(). Alberto. ___ --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] Thomson ST2030S and BLF
Andrew Joakimsen ha scritto: I too am wondering if someone has a contact at Thomson, some of the softkeys need to either be fixed or have the option to remove (like FwdVM and Pickup keys). In addition, has anyone notice a humming noise when using the handset? I can hear it and so can the person that I am calling. Honestly, I'm experiencing a good audio quality, no humming noise or hiss. Well, I'm using g711a... Alberto. ___ --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] Day/night service and indications on the phone
Olivier ha scritto:
I'm happy to report that with a very litte change to app_devstate.c
(just in the way ast_device_state_changed_literal() is called)
that module just compiles and works fine even without bristuffing
anything.
BTW I'm using a Thomson ST2030S phone with a status key subscribed
to a DS/xxx hint.
Thanks again for your precious help!
Could you elaborate ?
How is it working now ?
How you extensions.conf file looks like ?
Regards
Here's what I've got:
Configuration file for operator's phone:
...
[sys]
...
FeatureKeyExt10=S/sip:700
...
extensions.conf (within phone sip account's context):
...
;day-night service
exten = 700,hint,DS/night
exten = 700,1,DBGet(night=DEVSTATES/night)
exten = 700,n,GotoIf($[ ${night} = 2 ]?disable)
exten = 700,n,Devstate(night,2)
exten = 700,n,Playback(custom/night-service-on)
exten = 700,n,Hangup()
exten = 700,n(disable),Devstate(night,1)
exten = 700,n,Playback(custom/night-service-off)
exten = 700,n,Hangup()
(I have of course my own audio files that prompt
the operator about night service status)
The operator turns on/off the night service by just
pressing the F10 key on the phone, and its led
adjusts accordingly.
As to app_devstate.c, I've replaced any occurence of
ast_device_state_changed_literal(), which in bristuffed asterisk
takes 3 parameters (devname, cid, cidname) with
ast_device_state_changed_literal(devName)
as the original asterisk prototype requires (I don't care about
cid and cidname for this specific function).
To compile it outside bristuffed asterisk, just copy app_devstate.c to
the apps directory then edit the Makefile in it, adding
APPS+=app_devstate.so
after the first APPS= assignment.
I suggest you to do make and copy manually the resulting app_devstate.so
to your asterisk modules directory, instead of doing make install,
then issue a load app_devstate.so on the asterisk cli without restarting
it.
Thanks for the tip about thomson blf and firmware.
I'll try to trace sip dialog between thomson and chan_sip, although I'm
not very much into development. With some amount of luck I can try to
change the behavior of chan_sip code
Alberto.
___
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Re: [asterisk-users] Thomson ST2030S and BLF
Olivier ha scritto: Alberto, Call pickup is not implemented yet within Thomson ST2030 (1.50 firmware). More precisely, call pickup current implementation is not Asterisk compliant. A new release is scheduled for February (I've got this confirmed by Thomson 10 minutes ago) but we don't know if call pickup will be included. Regards I'd like to know what kind of compliance is required. I've tried to track what happens when a subscriber line key is pushed: - The Thomson phone sends an initial SUBSCRIBE message to Asterisk (each message is actually send twice, the first as anonymous, NACKed, the second with md5 digest auth, ACKed by asterisk) - The Thomson phone sends subsequent periodic SUBSCRIBE refreshing messages to Asterisk - When the SIP channel whose extension is hint-ed in extensions.conf gets busy/ringing/etc., Asterisk sends a NOTIFY message with a xml body containing the updated status on the line - The Thomson ACKs the NOTIFY and updates the LED status accordingly These steps work regularly. Now, when a line is ringing, if I press the flashing line key, the Thomson sends a SUBSCRIBE message to Asterisk instead of an INVITE (which is sent, on the opposite, when the line key is not flashing). Asterisk replies (I guess) correctly by ACKing and sending a NOTIFY (which is also ACKed by Thomson). Then nothing happens, the phone gives an error and...voilà, the key keeps flashing fast until next reboot. I wonder why the ST2030 sends a SUBSCRIBE upon key press when the key is flashing, while it sends an INVITE when the key is lit or off. Any clue on that? What is the ST2030 expecting back from Asterisk in order to proceed with call pickup?? It looks like the phone is NOT willing to send any pickup request... ___ --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] Thomson ST2030S and BLF
Olivier ha scritto: ... I didn't get any usable reply yet, beside usual maybe with next release. From http://bugs.digium.com/view.php?id=5014, I don't think one key call pickup is going to appear anytime soon with Asterisk. Hi Olivier. That's a pity. ST2030s is in my opinion one of the best SIP phones, with all features a phone needs (very good provisionig support, poe, double ethernet, line keys, subscriber keys, remote phonebook, audio quality...) compared to its low price. It would be so easy to issue an INVITE to the very same key extension and do the pickup via Pickup() dialplan function... Do you have a direct contact with Thomson guys? I've tried to reach them on e-mail or phone but with no success... Anyway, thanks again for the NOTIFY call-id patch tip. That's a new toy to play with for a couple of day before giving up. Alberto. ___ --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] Thomson ST2030S and BLF
Hello. Once again, I came up with a problem for which I can't seem to find a solution. I'm not able to make BLF work with Thomson ST2030 phones and Asterisk (1.2.13). I've set up hints in dialplan, as well as Subscibe keys on the phone. The LED status gets updated according to the associated line status. However, when a phone is ringing, If I try to pickup the call by pressing the flashing key on the Thomson phone, I get an error, and the key keeps flashing at high rate until I reboot the phone, even if the associate line goes back to idle. I'm using firmware 1.50t3. I've also patched chan_sip as indicated on this forum: http://www.ip-phone-forum.de/showthread.php?p=590842#post590842 No success. Any help would really be appreciated. Thanks, Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] Day/night service and indications on the phone
Hello everybody. I have created an extension that enables/disables the night service mode on asterisk (i.e. voicemail is started on incoming calls, instead of entering a queue). The night mode is activated/deactivated by the front desk operator when the office is about to close, by pressing a line key on the phone, which dials the related extension. Does anyone know a way to have an indication on the operator's phone using BLF-style hints or similar? I'd like to have a LED turned on when night mode is active (most traditional PBXes offer this feature). Is there a way, for instance, to force the device state for a dialplan hint, e.g. on a fake or local channel, so that I can map a BLF key on the phone to that hint? I have not find anything suitable so far (except for a PC system tray's icon, which is not applicable to my situation). Thanks, Alberto. ___ --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] Day/night service and indications on the phone
On Dom, Dicembre 17, 2006 15:26, Doug Lytle wrote: [EMAIL PROTECTED] wrote: Is there a way, for instance, to force the device state for a dialplan hint, e.g. on a fake or local channel, so that I can map a BLF key on the phone to that hint? [turn on mwi] touch /var/spool/asterisk/voicemail/context/device/msg0001.txt [turn off mwi] rm /var/spool/asterisk/voicemail/context/device/msg0001.txt -f Doug Thanks for the tip. But doesn't that conflict with the real message waiting indication? (The phone extension has its own voice mailbox). ___ --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] Day/night service and indications on the phone
On Dom, Dicembre 17, 2006 15:56, Michiel van Baak wrote: You can also use the devicestate commands in BRIstuffed asterisk. Michiel van Baak Thanks, this looks like what I need, although I'd better not to bristuff any of my asterisk boxes. I'll try to play with app_devstate.c alone (maybe it'll compile outside bristuff, without the need to patch the whole source). Alberto. ___ --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] Day/night service and indications on the phone
On Dom, Dicembre 17, 2006 16:10, [EMAIL PROTECTED] wrote: On Dom, Dicembre 17, 2006 15:56, Michiel van Baak wrote: You can also use the devicestate commands in BRIstuffed asterisk. Michiel van Baak Thanks, this looks like what I need, although I'd better not to bristuff any of my asterisk boxes. I'll try to play with app_devstate.c alone (maybe it'll compile outside bristuff, without the need to patch the whole source). Alberto. I'm happy to report that with a very litte change to app_devstate.c (just in the way ast_device_state_changed_literal() is called) that module just compiles and works fine even without bristuffing anything. BTW I'm using a Thomson ST2030S phone with a status key subscribed to a DS/xxx hint. Thanks again for your precious help! ___ --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] Siemens Gigaset SL75
Joao Pereira ha scritto: Do you know if it has 802.1x authentication as it is defined in EDUroam ( http://www.eduroam.org/ ) ? I never found a WiFi phone working with 802.1x I tested ZyXel Prestige 2000 but the sound was bad and it doesnt support 802.1x :( Thanks Joao Pereira Well, I have not tested 802.1x with eap/radius so far, but wpa/psk works quite well in a multi-AP roamed environment. The phone definitely has 802.1x authentication, at least that's what user guide and phone menus report. So far, this is the best wi-fi phone I ever put my hands on others phone I tested were (with my humble opinion on them): Zyxel P2000 - discrete but poor wi-fi options Linksys WIP300 - crappy, slow, freezing, battery lasts less than 30' talking UTStarcom F1000 - sufficient, bad display/menus, poor audio UTStarcom F1000G - ultimate crap, firmware is really bad, frequent disconnection from wifi net Samsung WIP6000 - good phone, but available in Italy only as Telecom Italia rebranded and locked plus it's a little too small, cellphone-like Nokia eSeries (60,70) - great smartphones but BAD BAD BAD sip stack and/or wi-fi integration, hoping for a fixing firmware update from nokia guys On the contrary SL75 has: - the right size for a cordless phone - a comfortable charging cradle - multilanguage interface (including Italian! yup!) quick reference guide - standard call functions (call waiting, hold, transfer 3-way conference) - a battery that lasts at least 2 hours talking (personally tested!) - soft plastic case which should prolong the phone's life with absent-minded employees, in case of dropping - good roaming features with manual setting of RSSI threshold - the best price/performance ratio What the phone really lacks is a TFTP-like automated provisioning capability. Unfortunately, as far as I know you can configure it only manually via web. I'll be testing radius eap-md5 in the next few days. ___ --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 IAXmodem HylaFAX
[EMAIL PROTECTED] ha scritto: Hi, I have a question: In my company a have a 100% IP Telephony implemented (Cisco Phones, Routers, Switches, etc). We are connected to our operator by two E1 lines. The PBX is on operator side (a H.323 Gateway is configured on call manager). I study the possibility to implement a Hylafax server. In some sites I read that with the IAXmodem is possible to have virtual modems, with don't so buy voip-fax-gateway or an expensive modem card with many channels. The how-to that I read is with Asterix. Is possible to use a HylaFax Server with IAXmodem and configure to work with Cisco Call Manager? Bests Regards, ___ --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 Theoretically... yes. In practice... good luck. I haven't tried T38 so far, but passing through g711 a/u-law from the remote side to hylafax makes nine fax transmissions out of ten to fail handshake. I have this: ISDN -- cisco 2811 --(sip)-- asterisk --(iax)-- iaxmodem --(ttydevice)-- hylafax/faxgetty (consider that iaxmodem is running on the same asterisk host, beign connected on 127.0.0.1 as a iax2 peer) It simply does not work, no matter how you set qos, precedence, priority, etc... It's really too easy for T30 frames to go out of sync. I would not recommend to spend even a single minute trying to set that up, I believe it's highly unreliable. Alberto. ___ --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] iax2 - wildiax phone myself puzzled
/support - state 5 (Unavailable) 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: Sending 9 on 4/20965 to 10.0.10.160:4569 2006-11-22 15:14:06 VERBOSE[6863] logger.c: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH 2006-11-22 15:14:06 VERBOSE[6863] logger.c:Timestamp: 9ms SCall: 4 DCall: 20965 [10.0.10.160:4569] 2006-11-22 15:14:06 VERBOSE[6863] logger.c:AUTHMETHODS : 1 2006-11-22 15:14:06 VERBOSE[6863] logger.c:USERNAME: support 2006-11-22 15:14:06 VERBOSE[6863] logger.c: 2006-11-22 15:14:06 VERBOSE[6863] logger.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ 2006-11-22 15:14:06 VERBOSE[6863] logger.c:Timestamp: 3ms SCall: 20966 DCall: 0 [10.0.10.160:4569] 2006-11-22 15:14:06 VERBOSE[6863] logger.c:USERNAME: support 2006-11-22 15:14:06 DEBUG[734] app_queue.c: Device 'IAX2/support' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. 2006-11-22 15:14:06 VERBOSE[6863] logger.c:REFRESH : 30 2006-11-22 15:14:06 VERBOSE[6863] logger.c: 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: New max nontrunk callno is 6 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: Creating new call structure 5 ... (the regreq/ack regauth sequence gets repeated many more times) then a lot of 2006-11-22 15:14:18 VERBOSE[6863] logger.c: Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ 2006-11-22 15:14:18 VERBOSE[6863] logger.c:Timestamp: 10011ms SCall: 4 DCall: 20965 [10.0.10.160:4569] Thanks in advance for any lead Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] gotoiftime and blocking calls
Tom Vile ha scritto:
I am trying to use the Gotoiftime CMD to not allow calls to be placed
between the hours of 12am-5am, except if you know the PIN number to
dial out and if the call is for 911.
What is the best way to implement this solutions?
I have the gotoiftime like so:
exten = s,1,GotoIfTime(5:00-11:59|mon-fri|*|*?custom-blacklist,s,1)
and using Read for the PIN like so:
exten = s,3,Read(Secret,,3)
exten = s,4,NoOp(${Secret})
exten = s,5,Gotoif($[${Secret} = 123]?6:8)
but I guess I am stuck at allowing 911 calls to go through and what
order to place them in.
Thanks for the suggestions.
Tom
The quickest way is to use two different extensions.
I don't know the rest of your dial plan (so this might conflict
with internal extensions), but it could work (of course you've to
replace whatever with your actual outgoing trunk channel:
[context-dialout]
exten = 911,1,Dial(whatever/${EXTEN})
exten = _X.,1,GotoIfTime(5:00-11:59|mon-fri|*|*?custom-blacklist)
exten = _X.,n(do_dial),Dial(whatever/${EXTEN})
exten = _X.,n,Hangup()
exten = _X.,n(custom-blacklist),Read(Secret,,3)
exten = _X.,n,NoOp(${Secret})
exten = _X.,n,Gotoif($[${Secret} = 123]?do_dial)
exten = _X.,n,Playback(sorry-dude-youre-not-allowed)
exten = _X.,n,Hangup()
Alberto.
--
--
Alberto Pastore
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P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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Re: [asterisk-users] Ringing a group of phones but not if they are busy
Chris Bagnall ha scritto: I need to ring a group of 8 phones, but not if they are already on another call. How can I determine which of those 8 phones are busy so I only ring the others? I've done this in the past by disabling call waiting on the phones and put all 8 phones into a ringall queue. Then, when you call that queue, the phones already on calls return SIP BUSY,whilst the others ring as normal. It's not perfect, but for most of our users the call waiting noise in the earpiece is an annoyance anyway. Hope that helps. Regards, Chris If you disable call waiting, then you don't need a queue. With grandstream gxp-2000 phones, calling Dial(SIP/phone1SIP/phone2SIP/phone3) rings only off-hook phones. However, I have also SPA-941 phones. Is it possible to disable the call waiting feature on Linksys SPA-941? I haven't succeeded so far... and the multiple Dial() method or the Queue are not working either. I had to change my extensions.conf macro to do ChanIsAvail sequentially, that is, for each phone I call ChanIsAvail and then check the results to see if the phone is busy. If not, I add it to the dialstring to pass to Dial() eventually. Since there are 10 phones to check, and the process is not atomic, it can (very rarely) occur that a phone is included in the dialstring but has just become busy, and the user gets the annoying call waiting tone. Any clue? -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] spc.exe
Sipura Profile Compiler is only for ITSPs and agreements does not permit that Regards Andrew Joakimsen escribió: Does anyone have a copy of spc.exe they could send me? ___ --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] Nokia E70
Michiel van Baak ha scritto: Hi, Anyone here has any experience with the Nokia E70 and asterisk ? I read on the nokia website this phone is capable of talking SIP and do Presence based on SIP/SIMPLE. Please share your experience, I'm thinking of getting one but want to be sure I can use it with * before I do. Thnx. Hi. we're using a couple of e70 and six e60 in our company. If you don't set Internet Phone as the primary connection, it works quite well (if you set Internet Phone as the primary choice, you'll have to reboot the phone often, at least once a day... that's what happens on all our phone even after the last firmware upgrade). We don't use Presence, however we can receive calls on our extension numbers, as well as make calls via wifi with good quality. We're using iLBC codec, since g729 was not that good. My own e70 is also set to connect to my home wifi access point and automatically register on my company's asterisk, so that I can make/receive calls when I'm home as if I were at office. Battery life is excellent, compared to the enabled radio functions (bluetooth is on for wireless headset, wi-fi is on, umts/gsm of course is on), about 1 day 1/2 with many phone calls. My greatest concern is about its poor exterior robustness... I hope I'll never let it fall to the floor, it looks like it'll break into a thousand pieces. I can say I'm overall satisfied, although I'm hopefully waiting for a more stable firmware. Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] some questions about atxfer usage
Antonio Almodóvar ha scritto:
Hi all.
I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)
When I transfer a call to an extension:
- The extension rings during 15 seconds and the call returns to the
transferer. Is there any possibility to recover the call before the
timeout of 15 seconds expires?
I mean, I would like to personalize the way of making transfers using
the feature of atxfer. How can I do that?
Thanks in advance.
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Hi Antonio.
Taking a look at the following code line from res_features.c:
newchan = ast_feature_request_and_dial(transferer,
Local,
ast_best_codec(transferer-nativeformats),
dialstr,
15000, // ---
outstate,
cid_num,
cid_name);
I assume that 15000 msecs is a hardcoded value...
You might want to replace it with some variable taken from
pbx_builtin_getvar_helper() results
but it involves recompiling at least the res_features.c module;
something more or less
like this (I haven't tested it!!!):
//these two lines go at the beginning of the if {} block
char *transfer_timeout_str;
int transfer_timeout = 15; //default value
//these lines replace the newchan = ast_feature_request_and_dial(...) one
//read the value (if any) from TRANSFER_TIMEOUT
//can be set in extensions.conf's [globals] (TRANSFER_TIMEOUT = 30)
transfer_timeout_str = pbx_builtin_getvar_helper(transferer,
TRANSFER_TIMEOUT);
if (transfer_timeout_str) {
transfer_timeout = atoi(transfer_timeout_str);
//sanity check
if (transfer_timeout = 0) transfer_timeout = 15;
}
newchan = ast_feature_request_and_dial(transferer,
Local,
ast_best_codec(transferer-nativeformats),
dialstr,
transfer_timeout * 1000, // ---
outstate,
cid_num,
cid_name);
Bye,
Alberto.
--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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Re: [asterisk-users] config template for Grandstreams
Todd- Asterisk ha scritto: I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate the config file from a template, but the templates posted on their website are for an old version of the phone firmware. Anyone have a tool or access to templates for the latest firmware versions? I guess the procedure is to modify the template, then run the configuration tool on the template to generate the specific downloadable file..? Thanks Todd ___ --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 Todd, afaik there's no updated template around. However you can update it yourself (with a bit of patience): just browse the web configuration pages for a gxp or bt phone, look at the source html on each page and figure out all the newer parameters (you'll see that the html form field names match the text template ones, for example P201, P43, etc). Write them down to the template, they'll work. Unfortunately it looks like grandstream phones are not able to download a commontemplate and a mac-address specific one, but just the latter one. You may want to write a small script that creates the cfgmac_addr.txt files (one for each phone) deriving it from a template, then compile them using the configuration tool provided by Grandstream. I manage about 250 phones (including bt200, ht2xx/4xx and gxp2000) with this method. My template still lacks the last additions (xml phonebook/screensaver, daylight savings, etc.) as I could not find the time for updating it yet. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] monitor-join does not seem to work.
Despite of monitor-join being equal yes, I get individual -in and -out files for queue calls. My box runs Asterisk 1.2.10 and I've set up real-time queues. Does anybody have any idea of what is going on? Thanks in advance. Carlos. ___ --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] Zaptel 1.2.11 released
Asterisk Development Team ha scritto: The Asterisk Development Team is pleased to announce the release of version 1.2.11 of Zaptel. Where is it??? The link on asterisk.org is broken... Also, no Changelog anywhere. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] Thomson ST2030 and Asterisk BLF
Hi everybody.
I know there have been some posts in the past about this subject.
However it seems I cannot get the Thomson ST2030 phone
to work with BLF and call pickup.
Firmware on the phones is 1.5t3
I've applied then patch to chan_sip.c which adds the else condition
} else if (strstr(p-useragent, THOMSON)) {
p-subscribed = DIALOG_INFO_XML;
somewhere in the handle_request_subscribe() function.
The hints are properly configured as well as the subscribecontext in
sip.conf/extensions.conf
In fact, the busy lamp is working (I can see
busy lines and ringing lines on the phone),
however call pickup is not.
When a line key is flashing (i.e. the associated sip phone is ringing),
if I press that key, the phone sends a SUBSCRIBE sip message
to asterisk.
I don't understand exatctly what the phone is expecting back
from asterisk or how asterisk handles the SUBSCRIBE message,
however the call is *not* picked up, and the status line key
gets fast-blinking, and remains in that status,
being unusable, until I reboot the phone.
Any hint?
Thanks.
Alberto.
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Re: [asterisk-users] [OT] wi-fi ip phone scenario
Alban ha scritto: Yes, same channel and same ESSID for all AP's. Are you connecting each AP to the LAN? Or only one connected, and the others as relay? With WDS, you have to keep same channel and ESSID for a good roaming. If connected to the lan, doing it worked really good for me, roaming was working in the same way as with WDS (no latency). No WDS, all APs hardwired. ___ --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] Asterisk and Siemens C450IP
Hi. Again one big mysterious problem I hope some good guy can help me solve. I'm trying to connect some Siemens C450 SIP IP Dect phones to asterisk (1.2.13) (I have actually 3 handsets + 3 ip base). After configuring them and rebooting, all of them register properly on asterisk, then, after the first call, they appear no more registered as registered in asterisk, and on the handset the display shows SIP registration failed. Has anyone got the same problem? Has anyone ever tried to operate more than one C450IP in the same open space? fyi: - there are no lan/ethernet problems - I've tried with different qualify values (0, 2000, 5000) - no nat involved - everything is running on a private lan - phones reply to ping even when qualify times out - any other sip phone (wired to the lan) works just fine It looks like the rise of the machine... every new phone I try drives me crazy. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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 7960 Skinny calling SIP phone
Well, I've never actually been able to make chan_skinny work with 79xx
phones.
I found the chan_sccp to work quite well:
http://chan-sccp.berlios.de/
plus this patch for a problem on MeetMe (I don't remeber where I found
it, but it works!):
diff -uNr chan_sccp-20060408.org/sccp_pbx.c chan_sccp-20060408/sccp_pbx.c
--- chan_sccp-20060408.org/sccp_pbx.c 2006-04-08 14:20:17.0 +0200
+++ chan_sccp-20060408/sccp_pbx.c 2006-05-17 17:14:15.0 +0200
@@ -290,6 +290,12 @@
static int sccp_pbx_answer(struct ast_channel *ast) {
sccp_channel_t * c = CS_AST_CHANNEL_PVT(ast);
+ // if channel type is undefined, set to SCCP
+ if (!ast-type) {
+ sccp_log(1)(VERBOSE_PREFIX_3 SCCP: Channel type
undefined, sett
ing to type 'SCCP'\n);
+ ast-type = SCCP;
+ }
+
if (!c || !c-device || !c-line) {
ast_log(LOG_ERROR, SCCP: Answered %s but no SCCP
channel\n, as
t-name);
return -1;
I recommend using SIP firmware anyway... the conversion process is a bit
annoying but
as far as now 7940/7960 are really stable IP phones.
I am currently using chan_sccp only for 7902 phones (I've just got 2 of
them)
which do not support SIP firmware.
Will Roy ha scritto:
Before I got down the path of converting a Cisco 7960 I have over to
SIP I wanted to try and set it up using Skinny.
The phone registers ok with Asterisk. When I call a SIP softphone
extension on my network the call is made and I can answering it.
However no voice is heard over the call.
When I debug Skinny on the console after the call has connected I see
the following messag:
Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7]
What additional information would be required to troubleshoot this? or
should I stop wasting time and just convert the phone to SIP? :)
regards
Wil
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--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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Re: [asterisk-users] Diva server 4bri and Portuguese BRI
Pedro Silva ha scritto: Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --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 I'm not sure about Portuguese operators standard, but I bet ETSI-DSS1 should work just fine. The interface mode is surely TE. The DID/MSN should not affect outgoing calls, I generally leave DID off unless the telco company has that service active. If you're using the diva server for linux package from eicon (divas4linux, currently rel. 8.2), you should find a very simple utility named telsampl under /usr/lib/eicon/divas which you can run besides asterisk, to test outgoing calls. You should run it with this command line: telsampl -c x where x is the bri port you wish to test (1..4) then at the prompt type c and enter a pstn number, e.g. your mobile phone, then you can watch the log onscreen. If the outgoing call works, then your isdn setup is correct, and the problem is in asterisk. The message from xlite is not meaningful, as it could occur on many situations. You should watch the debug output on asterisk console. That helped me a lot. Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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: [OT] wi-fi ip phone scenario
Martin Joseph wrote: I think it's cleary true that wiring WIFI infrastructure is easier and more reliable then WDS. On the other hand, I have been running my little network with WDS for over three weeks now, and it has been completely reliable. The tricks where to configure things properly and to have the bases closer together then one would think would be needed. Once this was setup. It works, and it keeps working. We had a couple of stress tests also, one black out and one unplugged router (carpenter). Came up cleanly and continued working fine. No mis-registrations and no problems. Marty Can I ask you guys which phones are you using? Alberto. ___ --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] Diva server 4bri and Portuguese BRI
Marco Mouta ha scritto: pls post your misdn.conf as well as extensions.conf May be i can help. Sou Português:) On 10/29/06, *Pedro Silva* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Thanks Alberto! I tested with telsampl like you said (with various configurations for de diva) and this not works...:( The trace is: Enter destination address: 273xx --Conn_Req(273xx) Connect_Con-- [29]:Disc_Ind-- --Disc_Res **Call cleared*** Any idea for the possible problem? Thanks and best regards, PS. I think Pedro is not using mISDN, he's using chan_capi, but his problem arises before asterisk is involved. Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.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] [OT] wi-fi ip phone scenario
Alban ha scritto: I've made some tests with Hitachi WIP3000 and 5000, works really good with roaming (without authentification). Some parts of the AP in the mesh are wired (no WDS), some others are not (using WDS), but all use the same SSID and channel. In all cases roaming was fast, quite not possible to hear it. Besides, with UTstarcom, roaming in the same mesh was not working well. Hope it helps Alban So you're all using access points on the same channel? I've been told that to make a good roaming wifi lan access points must have coverage areas overlapping with the next ap for a good 20%-25% of it (just to let the client roam to another AP while moving), and they should use different channels according their topology, in order to minimize adiacent channel interference, for instance something like this 1--13-4 | | | | | | | | | 10-6--9 | | | | | | | | | 8--12-2 I'm confused... ___ --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] ISDN-BRI issue
Tzafrir Cohen ha scritto: Works fine with Junghanns' cards. One simple thing for you to test: set one port in TE mode and one port in NT mode (move all 5 jumbers of that port to the other position to get it into NT mode). Then try to make a loopback connection (using a standard ethernet cable). Here you control both ends and thus there are less configuration pains. One thing that could be wrong is if both sides do not agree on the line settings. Where do you connect to? What do you have on zaptel.conf ? On zapata.conf? The very first test I did was to set a loopback between two spans (one in nt mode, the other in te mode). I wrote a small script to setup calls between the span continuously, and... guess what? I let the system run for 4 hours, place about 20.000 calls with no problem at all. Unforutnately, as soon as I connect the spans in te mode to Telecom Italia's NT1 lines, after a random time (from 15 minutes to 1 hour) and a random number of outgoing calls (the first 30 calls gets routed with no problem), one or more spans begin reporting Layer 1 Down. I tried to do a bri intense debug span x, all I see are SABME packets sent from the card to the NT1 line, with no UA reply, but INCOMING CALLS ARE WORKING ANYWAY!! The most strange thing at all is to be on the phone speaking with someone, on the very same line which bristuff is complaining about, in the very same moment in which bristuff is reporting layer 1 down... This does not happen only on my line. I saw it with my own eyes on 3 different asterisk boxes with 3 different Junghanns quadBRI, on 3 different PC (IBM/HP), on 3 different sets of ISDN lines (p2p, p2mp...) Just fyi, here's my really simple zaptel/zapata config (btw, I think I've tried all n! permutations of parameters for over two weeks...): bristuff 0.3.0-pre-1v, asterisk 1.2.13, zaptel 1.2.10, libpri 1.2.4 zaptel: loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 span=3,1,3,ccs,ami bchan=7-8 dchan=9 zapata: language=it switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown resetinterval=never priindication=outofband callprogress=yes usecallingpres=yes echocancel=yes echocancelwhenbridged=yes echotraining=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 context=ingresso signalling=bri_cpe channel = 1-2 channel = 4-5 signalling=bri_cpe_ptmp channel = 7-8 -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] wi-fi ip phone scenario
Andrew Joakimsen ha scritto: Are you using WDS? While it won't totally fix every issue, I've found in my trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but re-registartion won't require a reboot). Nope. I've a group of 12 indoor colubris access points + 2 outdoor ones, and a msc5200 controller unit (the area is quite wide with two 4-storey buildings, two 8 sq.feet hangars, one 4 sq.feet outdoor parking lot) no wds, all of them are wired to ethernet switches, no wireless bridging. Diversity on all APs, automatic transimt power and channel selection (that seems to work when monitoring devices). I've tried linksys wip300/wip330, nokia e60/e70, utstarcom f1000/f1000g, samsung wip6000. The main problem (apart from firmware bugs/crashes which I hope should be fixed on newer versions) is that phones tend to stick to their associated AP even when it's clearly time to move to the next AP: if you watch the phone's rssi indicator while you walk inside the coverage area, you can see the value decreasing to almost no signal before reassociating, which is unacceptable. The only phones which seem to deal very well with it are the nokia eSeries. Unfortunately they have many other major issues. (two weeks ago I gave 3 firmware-updated e60 to my bosses to replace their cellphones and after one week they almost threw them back to me, complaining about the fact they had to reboot the phone at least twice a day because it freezed...) Also, on asterisk the qualify=2000 sip setting seems to be to low, as the console shows repeated LAGGED/UNREACHABLE/REACHABLE notices on those phones. From this experience I think wi-fi technology is not really mature enough to replace dect right now. Maybe I'm wrong. At least I hope so, my bosses are not really happy with the new system. ___ --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 and ISDN and Hylafax
Hi. I have now many customers using hylafax + asterisk, and all of them have proven to be reliable. Two are using diva server 4bri 8m + CAPI (asterisk here is not involved, as the incoming fax call gets directly to ttyds0x devices, and the numbers assigned to the lines by our telco are excluded from extensions.conf), which is absolutely *perfect*, not even a single miss (except of course for remote party's troubles). Another couple of them is using a Digium T110 pri card with asterisk+iaxmodem (0.1.14) on the same box, and they work just fine (the only limitation is running at 9600 instead of 14400 bps) The latter two boxes have 30 software instances of iaxmodem (ttyIAX1..ttyIAX30) and, although we've never tested what happens with cpu (P4 xeon 2.8 uniproc) when all of them are DSPing, the server average load is 500 incoming faxes/day, with peaks of 6-7 simultaneous incoming fax jobs. Bye, Alberto. Thomas Winter ha scritto: Am Thursday 26 October 2006 23:35 schrieben Sie: On Thu, 26 Oct 2006, Thomas Winter wrote: I would recommend the Eicon DIVA Server 4BRI cards. They have a capi interface which is used by chan-capi (chan-capi.org) and onboards DSPs for the faxing. You can use this for send and receive faxes and/or use capi4hylafax in parallel with asterisk/chan-capi. sounds good, you think it will run reliable? ___ --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 -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] ISDN-BRI issue
Frédéric Blaise ha scritto: Hello all Asterisk 1.2.10 BRIstuff PRE-1s Debian sarge I have an issue with my ISDN-BRI line. Asterisk tells me my D-Channel is down, no matter is I have an actual line plugged in or not. == Primary D-Channel on span 1 down Try with signalling=bri_cpe even if your lines are set as point to multipoint, at least that should make your card trying to keep layer 1 up, even if this won't probably solve it. As a matter of fact I'm getting to conclude that bristuff + hfc-4s card is not working whatsoever. I believe there's something wrong in the way bristuff manages layer 1 (not sure if that's a driver problem, hardware problem or both). Alberto. ___ --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 and ISDN and Hylafax
Olivier ha scritto: What about telephony features using chan-capi and Asterisk ? Are those features on par with msidn+Asterisk or bristuff+Asterisk (maybe I'm mixing up things together) ? Cheers I'm running my own company's pbx with diva 4bri, diva server for linux 8.2, chan_capi from melware.org and everything is working just fine. I was able to use early B3 connect and everything else related to CLID. I have two isdn lines as point-to-multipoint and one as p2p. One isdn line is shared between asterisk/chan_capi and hylafax via diva's tty interfaces. Everything has been perfectly working since november 2005, when we first started this new pbx as a replacement of the old Samsung DCS, we almost forgot about its existence, as we did never have to put hands on it to fix problems (except for some ordinary maintenance and diva server software upgrades). Alberto. ___ --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] [OT] wi-fi ip phone scenario
Hello everyone. I know it's a little bit off-topic, but I was just wondering... Has anyone ever had any experience with asterisk, a wi-fi meshed lan (with more than one access point) and wi-fi sip phones? I made some tests but I'm not really satisfied Wi-fi phones are a curse (as far as I know even Nokia eSeries -I personally own an e70 model- have their flaws): - random sip registration failures - ridiculous battery life - bad audio quality even with optimal radio environments - crashes, system freezes - ... - slow responsiveness to asterisk qualify pings (OPTIONS) but I can even live with that. The major problem is... roaming between cells. Is that a dream or something that can actually work? Unfortunately I have to replace a good old DECT network (I know it'll never compare to DECT)... Alberto -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] Junghanns quadBRI and mISDN
Hi. I'm trying to run a Junghanns quadBRI card with mISDN drivers. I'm able to compile kernel mode user mode mISDN components as well as chan_misdn. The misdn-init config properly detects the card and starts the hfcmulti driver; lsmod shows all required drivers are loaded. However, the misdnportinfo seems not able to find any card. Has any one successfully managed to run Junghanns cards with mISDN? (there are a couple of serious issues using bristuff and we've been looking for alternate drivers). Thanks, Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] Junghanns quadBRI and mISDN
Maybe I found the cause... My Junghanns quadBRI PCI subsystem ID is 0xB552 (that is, quadBRI version 2.0), while mISDN expects 0xB550 (quadBRI version 1.0) I'm wondering what differences lie in the two boards from a driver's perspective... I'll try to recompile mISDN by adding also subsys=0xB552 to the list of supported pci devices. I'm not very familiar with kernel drivers, so... good luck to me. Alberto. Alberto Pastore ha scritto: Hi. I'm trying to run a Junghanns quadBRI card with mISDN drivers. I'm able to compile kernel mode user mode mISDN components as well as chan_misdn. The misdn-init config properly detects the card and starts the hfcmulti driver; lsmod shows all required drivers are loaded. However, the misdnportinfo seems not able to find any card. Has any one successfully managed to run Junghanns cards with mISDN? (there are a couple of serious issues using bristuff and we've been looking for alternate drivers). Thanks, Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] Problems with chan-capi and Eicon Diva 4BRI
Hi Klaus. I'm not sure about the timer expiry meaning, but you could use the xlog command (usually found in /usr/lib/eicon/divas) Just run it as root indicating which span (1..4) you want to trace: ./xlog -c 2 that shoud show you layer 1 layer 2 dump Alberto. Klaus Darilion ha scritto: Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show incoming (PSTN-Asterisk-PBX) calls, no outgoing. Thus I suspect that the Asterisk--PBX link was broken. In the Asterisk message file I only see Recovery on timer expiry errors, like below: Oct 20 17:18:18 VERBOSE[19772] logger.c: == ISDN2#02: Incoming call '347x' - '32xx' Oct 20 17:18:18 VERBOSE[19772] logger.c: -- ISDN2#02: Updated channel name: CAPI/ISDN2/32xx-8ab6 Oct 20 17:18:18 VERBOSE[2663] logger.c: -- Executing Dial(CAPI/ISDN2/32xx-8ab6, CAPI/g2//b|90) in new stack Oct 20 17:18:18 VERBOSE[2663] logger.c: -- Called g2//b Oct 20 17:18:19 VERBOSE[19772] logger.c: -- ISDN2#02: Updated channel name: CAPI/ISDN2/32xx11-8ab8 Oct 20 17:18:19 VERBOSE[2663] logger.c: -- ISDN4#02: Updated channel name: CAPI/ISDN4/1-8ab9 Oct 20 17:18:19 VERBOSE[2663] logger.c: -- ISDN4#02: Updated channel name: CAPI/ISDN4/11-8aba Oct 20 17:18:26 VERBOSE[19772] logger.c: ISDN4#02: CAPI INFO 0x34e6: Recovery on timer expiry Oct 20 17:18:26 VERBOSE[2663] logger.c: == ISDN4#02: CAPI Hangingup for PLCI=0x104 in state 4 Oct 20 17:18:26 VERBOSE[2663] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Oct 20 17:18:26 VERBOSE[2663] logger.c: -- Executing Hangup(CAPI/ISDN2/32xx11-8ab8, ) in new stack Oct 20 17:18:26 VERBOSE[2663] logger.c: == Spawn extension (frompstn, 32xx, 2) exited non-zero on 'CAPI/ISDN2/32xx11-8ab8' Oct 20 17:18:26 VERBOSE[2663] logger.c: == ISDN2#02: CAPI Hangingup for PLCI=0x202 in state 7 Oct 20 17:18:26 VERBOSE[19772] logger.c: ISDN2#02: CAPI INFO 0x34e6: Recovery on timer expiry What does the timer expiry exactly mean? Was it a Layer2 or Layer 3 problem? How can I find out more or how can I activate more BRI debugging for the case it happens again? Are there any known problems? We are using: Asterisk 1.2.12.1 chan_capi-0.7.0 divas4linux-melware-3.0.3-106.650-1 Diva Server 4BRI-8M 2.0 PCI Thanks Klaus ___ --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 -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] vISDN, mISDN, bristuff [was: Re: Bristuff qozap drivers problem]
-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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: ZapHFC quadBRI D-Channel going down randomly
I tried the same, and my Telco company told me (although sometimes it's hard to trust them, you never know what kind of guy from the call center is answering your call) that p2p lines already have l1 permanent. Nonetheless it goes down sometimes for quite long periods. I'm starting wondering whether it's some sort of kernel related problem (i.e. irq sharing settings etc.) by which the card loses packets. Henrik Woffinden ha scritto: I have the exact same problem on a normal ISDN2 BRI line. I solved it by having my Telco put layer 1 to permanent. Best regards, Henrik Woffinden Alberto Pastore wrote: asterisk ha scritto: On most traditional pabx's it's possible to set layer 1 to permanent or call. It sounds like your system is configured for permanent and your lines to call. How you would set this on asterisk I have no idea. fadge The question is: is it possible I am the only one with such problems on all asterisk boxes on different sites and different ISDN lines? I've googled around on many forums but no one seems to have this one. The old replaced PBXs had layer 1 set for call, as you say, and they showed no problems at all. With asterisk as a PBX, every 2-3 hours, you cannot dial out for 5 to 15 minutes then everything gets back to normal (no idea about what triggers the return to working state). ___ --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] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M
If you have divas4linux package installed (from Eicon), you can use the Config textual gui utility, it always reports which cards (model and revision) are found in your system. Klaus Darilion ha scritto: Hi (Armin)! Does someone knows how to identify the type of the card? The delivery note says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M. What is it really? Are there any Eicon tools to identify the card type? thanks klaus :0a:03.0 Network controller: Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 (rev 01) Subsystem: Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32, Cache Line Size: 0x10 (64 bytes) Interrupt: pin A routed to IRQ 77 Region 0: Memory at fdeffc00 (32-bit, non-prefetchable) [size=256] Region 1: I/O ports at cc00 [size=256] Region 2: Memory at fc00 (32-bit, non-prefetchable) [size=16M] Region 3: Memory at fdee (32-bit, non-prefetchable) [size=64K] Capabilities: [40] Power Management version 1 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [48] #06 [0080] Capabilities: [4c] Vital Product Data ___ --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 -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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: ZapHFC quadBRI D-Channel going down randomly
asterisk ha scritto: On most traditional pabx's it's possible to set layer 1 to permanent or call. It sounds like your system is configured for permanent and your lines to call. How you would set this on asterisk I have no idea. fadge The question is: is it possible I am the only one with such problems on all asterisk boxes on different sites and different ISDN lines? I've googled around on many forums but no one seems to have this one. The old replaced PBXs had layer 1 set for call, as you say, and they showed no problems at all. With asterisk as a PBX, every 2-3 hours, you cannot dial out for 5 to 15 minutes then everything gets back to normal (no idea about what triggers the return to working state). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Pastore Sent: 16 October 2006 17:26 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ZapHFC quadBRI D-Channel going down randomly Hi. I'm running some asterisk boxes on different sites, some equipped with a couple of ZapHFC cards, others with Junghanns quadBRI cards. All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6) and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with kernel 2.6.17.3 The cards are connected to Telecom Italia's NT1/NT1+ S/T lines; some of them are point-to-point, others are point-to-multipoint. I keep getting always the same problem: after some hours of regular working, some boxes report the usual message Primary D-Channel on span n down (where n is different every time, depending on the number of active bri spans) I've read on previous postings that having layer 1 down on ptmp spans is normal. However after getting a down message (on ptp spans too!) I'm no more able to place outgoing calls on that span, until I restart asterisk zaptel drivers. Sometimes, they get back working by themselves (with the related span up notification) after a random time period. During the down period, incoming calls are regularly served. However these calls do not change the status of the span, i.e. as soon as the calls are hung up, the span gets down again. I've tried to capture the dialog between the card and NT1 equipment, and during the down state, I got this repeated over and over: Sending Set Asynchronous Balanced Mode Extended [ 00 8b 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 069EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] == Primary D-Channel on span 1 down In zapata.conf I'm pretty sure I've always set the correct signalling settings (switchtype = euroisdn, signalling = bri_cpe_ptmp or bri_cpe depending on the case) In /etc/zaptel.conf, I've tried many combinations with no difference; my current settings are like this: span=1,1,0,ccs,ami bchan=1-2 dchan=3 span=2,1,0,ccs,ami bchan=4-5 dchan=6 etc Any clue? Thanks, Alberto -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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 -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] ZapHFC quadBRI D-Channel going down randomly
Hi. I'm running some asterisk boxes on different sites, some equipped with a couple of ZapHFC cards, others with Junghanns quadBRI cards. All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6) and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with kernel 2.6.17.3 The cards are connected to Telecom Italia's NT1/NT1+ S/T lines; some of them are point-to-point, others are point-to-multipoint. I keep getting always the same problem: after some hours of regular working, some boxes report the usual message Primary D-Channel on span n down (where n is different every time, depending on the number of active bri spans) I've read on previous postings that having layer 1 down on ptmp spans is normal. However after getting a down message (on ptp spans too!) I'm no more able to place outgoing calls on that span, until I restart asterisk zaptel drivers. Sometimes, they get back working by themselves (with the related span up notification) after a random time period. During the down period, incoming calls are regularly served. However these calls do not change the status of the span, i.e. as soon as the calls are hung up, the span gets down again. I've tried to capture the dialog between the card and NT1 equipment, and during the down state, I got this repeated over and over: Sending Set Asynchronous Balanced Mode Extended [ 00 8b 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 069EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] == Primary D-Channel on span 1 down In zapata.conf I'm pretty sure I've always set the correct signalling settings (switchtype = euroisdn, signalling = bri_cpe_ptmp or bri_cpe depending on the case) In /etc/zaptel.conf, I've tried many combinations with no difference; my current settings are like this: span=1,1,0,ccs,ami bchan=1-2 dchan=3 span=2,1,0,ccs,ami bchan=4-5 dchan=6 etc Any clue? Thanks, Alberto -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] WRT54GP2 provisioning
If you are an ITSP provider, you could do with SPC tools (provided by Linksys to ITSPs) Regards Curt Shaffer escribió: Can anyone point me to a good source for provisioning WRT54GP2 from a central server? Thanks Curt ___ --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 -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] Google talk and Asterisk 1.4
Check it! http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk Robert LaPoint escribió: Hello All Does anybody know where I can find information on configuring Asterisk 1.4 to work with Google talk. ___ --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] Google talk and Asterisk 1.4
What did not work? I made test under SVN Trunk and only have issues with audio behind NAT clients. You could check at bugs.digium.com Gtalk development state and bugs resolved. I did not make test with 1.4 beta 2 , so i could not help you more Regards Robert LaPoint escribió: I have already tried to follow this document but it did not work under 1.4, so I am just wondering if Google talk is even supported under asterisk 1.4 yet. Thanks Alberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Saturday, September 30, 2006 3:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google talk and Asterisk 1.4 Check it! http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk Robert LaPoint escribió: Hello All Does anybody know where I can find information on configuring Asterisk 1.4 to work with Google talk. ___ --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] señalizacion te110p, signaling te110p
Maybe you could try an asterisk forum in spanish in order to get better results using your native language. DiegoF escribió: hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, bueno como habia dicho quiero conectar una pbx a una te110p, la pbx me ofrece señalizacion r2 europea en cable rj45 o coaxial. ese tipo de señalización me sirve para la tarjeta te110p, ademas, alguno de esos dos tipos de conexiones me sirven o tengo que comprar algun adaptador. vi algo que tenia que usar un balum, es necesario para cualquiera de las dos conexiones?. cual tipo de conexioon me recomiendan mas? necesito saber algo mas sobre la pbx para configurar en la te110p? atentamente diego fernando güiza arce / hello to all, I have a doubt, ye I have solved some but others arrive, good since te110p had said I want to connect a PBX to one, the PBX offers señalizaciòn to me r2 European in cable rj45 or coaxial that type of signaling is used for the card te110p to me, in addition, some of those two types of connections serves to me or I must buy some adapter. I saw something that tapeworm that to use a balum, is necessary for anyone of the two connections. as type of connection they recommend to me but? I need to know something but on the PBX to form in te110p? kindly diego fernando güiza arce // -- // DiegoF // // Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados // // Se han fijado que cuando estan solos...no hay nadie??? // // Cada vez que me siento a pensar, lo unico que consigo es sentarme. // ___ --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] Linksys SPA400
It has a proxy inside (asterisk), you could register to it as a regular sip proxy, so you could use it. Carlos Chavez escribió: Does anyone know if the Linksys SPA400 is compatible with Asterisk or is it only for the SPA9000 system? It is interesting because it is a 4 FXO ATA at a reasonable price. ___ --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] Linksys SPA400
Not True! You could register against it any spa product, and also asterisk. Cory Andrews escribió: It's only designed for use with the SPA-9000 (LVS-9000) product ecosystem. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Thursday, September 21, 2006 1:00 PM To: Asterisk Subject: [asterisk-users] Linksys SPA400 Does anyone know if the Linksys SPA400 is compatible with Asterisk or is it only for the SPA9000 system? It is interesting because it is a 4 FXO ATA at a reasonable price. ___ --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] Attended Transfer Asterisk 1.2.11
Im updating from 1.2.9.1 to 1.2.11 and im having a issue with attendad transfer via SPA 941 that i did not have with 1.2.9.1. I get this message on Cli log. Sep 14 16:09:52 NOTICE[5780]: chan_sip.c:6897 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]'. Both legs must reside on Asterisk box to transfer at this time. I have canreinvite=yes on all extensions, and tried with canreinvite=no, but same happens. When i press tranfer, i could talk with destination extension, but when i transfer call, it hangups both sides. Regards -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] Choppy MOH (Cisco gateway)
VAD maybe was caussing this. Regards Zeeshan Zakaria escribió: Actually the problem was somewhere in the Cisco equipment, as the service provider has confirmed. Some option in their device to conserve bandwidth by compressing voice data was causing this choppyness. As they've turned this option off now, MoH works perfect. ___ --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 logging per day
You could use logrotate or you could configure your cron to send asterisk -x logger rotate, which it will do what you want. Regards Christophorus Laube escribió: Hi list, I am searching for a possibility to let my * log per day. So that a new logfile is taken every night at midnight, with the date in the file name. Is there a way to do so? Does anyone of you has tried that before? Regards, Christophorus ___ --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 -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] Whcih phones are better for mass deployment
I prefer Linksys ones. Spa 9xx series, are great, and provisioning from Sipura/Linksys is much better than PA1628 (Unencrypted). Supports https,tftp and http. With Encryption. Vonage use it. Regards Thomas Kenyon escribió: Michael Graves wrote: Polycom Aastra are both great in this manner. Even Cheapo PA168S phones will remotely update their configurations from a simple handful of files from a tftp, ftp or http url. (which is admittedly only simple to do with 30 phones simultaneously if you use PoE). I'm very surprised if a Grandstream can't manage 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 ___ --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] QUINTUM TENOR ASM200 Configuration
If you want to answer directly to him, try Reply to all, and delete [EMAIL PROTECTED] email address. It is not so had to do. FRANCISCO PEREZ-LANDAETA escribió: Hi, this message is for Steve. Sorry for replying to the digest. It wasn't my intention. I would appreciate if you can guide as to how make the tenor asm200 work with asterisk. I am using asterisk at home. I guess my problem is configuring the tenor so that it is recognized and can take calls from asterisk (both ways). If you can help me out and send me a sample config i would be very thankful. My config is an asterisk at home box, and i wish to be able to have my quintum register to the asterisk. Both devices will be in different lans. My intention is to be able to call my asterisk box (in my home), using my quintum box (in my office) and vice versa. thanks, Francisco _ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline ___ --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] Call Processing Slow 11 seconds
Yes you could script a dialplan putting ... and S0 (zero) at the end. An example : (xxS0) It will dial 6 digits directly when you enter the 6th. You could learn how to adapt your Linksys dialplan looking this wiki. http://voip.wikispaces.com/ [EMAIL PROTECTED] escribió: Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated. -- Original message -- From: G.Jacobsen [EMAIL PROTECTED] In case you use an adapter or voip phone: Did you try to press hash # after the number ? - then the adapter/voip phone dials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original Message *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] *Sent:* Samstag, 9. September 2006 15:15 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 Asunto: RE: [asterisk-users] Call Processing Slow 11 seconds De: G.Jacobsen [EMAIL PROTECTED] Fecha: Sat, 9 Sep 2006 17:20:05 + Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.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 ___ --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 to Install H323
I think remember there is a readme on /docs that talks about chan_h323.Check it ! Anyway you could try too at voip.info dot org. Regards Wasif escribió: Hello, Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 . Thanks Wazb ___ --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] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls
I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works fine. Are you canreinvite=yes ?. I have not been notice any problem related to transferring calls (blind and attended) Regards Dan Serban escribió: I have a system running Asterisk 1.2.10 (Debian packaging) and about 50 Linksys SPA-942 phones, after the initial config and mass deployment of the phones everything looks like it's configured well. When an incoming call is answered and then attempted to be xfer'ed via the soft button on the phone itself, it seems that if you hit the button twice in quick succession, there is no problem (effectively a blind transfer), if then I try to tell the other extension that Joe is calling to sell you a fridge and hit xfer, the calling party cannot hear what that person at the extension is saying. Sometimes the tables are fully turned, the caller can hear, but the operator can't hear a thing. One thing's for sure, if you hit the button quickly (blind transfer) it works no problem at all. This is what I see asterisk saying when I transfer the call unsuccessfully. == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' I've looked at the macro with a fine tooth comb, I cannot see any problems with it whatsoever, (though that doesn't mean that my ignorance isn't getting in the way). I found some mention on the digium mantis bug tracker, here's the link: http://bugs.digium.com/view.php?id=7421 Before I try and patch the source (which I'm hesitant to do since I run the debian packages), is there another solution or maybe an unidentified issue that I haven't been able to decipher? If there's more information that I can provide to solve this problem, I'd be happy to do so. Thank you. ___ --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] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls
I use canreinvite=yes in my config files, and it does work, so maybe its a spa 941 misconfiguration. I think if nat=no sometime it has problems if you are behind NAT, but under same network it must not fail. Which firmware are you running on spas? Dan Serban escribió: Alberto Sagredo wrote: I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works fine. Are you canreinvite=yes ?. I have not been notice any problem related to transferring calls (blind and attended) Thank you for your response, it gave me a nudge to check the configuration in the sip.conf file. It seems that if I set canreinvite=no for every SIP peer, it works! And I have found no other adverse effects. Strange issue... Regards Dan Serban escribió: I have a system running Asterisk 1.2.10 (Debian packaging) and about 50 Linksys SPA-942 phones, after the initial config and mass deployment of the phones everything looks like it's configured well. When an incoming call is answered and then attempted to be xfer'ed via the soft button on the phone itself, it seems that if you hit the button twice in quick succession, there is no problem (effectively a blind transfer), if then I try to tell the other extension that Joe is calling to sell you a fridge and hit xfer, the calling party cannot hear what that person at the extension is saying. Sometimes the tables are fully turned, the caller can hear, but the operator can't hear a thing. One thing's for sure, if you hit the button quickly (blind transfer) it works no problem at all. This is what I see asterisk saying when I transfer the call unsuccessfully. == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' I've looked at the macro with a fine tooth comb, I cannot see any problems with it whatsoever, (though that doesn't mean that my ignorance isn't getting in the way). I found some mention on the digium mantis bug tracker, here's the link: http://bugs.digium.com/view.php?id=7421 Before I try and patch the source (which I'm hesitant to do since I run the debian packages), is there another solution or maybe an unidentified issue that I haven't been able to decipher? If there's more information that I can provide to solve this problem, I'd be happy to do so. Thank you. ___ --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] Linksys SPA-3000 Administration Guide
This Guide is offered as i know only to ITSP and large distributors not to end-users. You could find a User Guide for SPA 3102 at Linksys Website. Regards Marcos Rubino escribió: Anybody have a recent copy of the Admin Guide (not the user guide) for the SPA3000/3102? The only one I was able to find was a terribly written two year old one on the Sipura site[1] and Linksys says you have to be a Service Provider to get one from them. [1]http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf I am just a humble VOIP enthusiast, can anybody hook me up? Please CC me on the reply (or respond directly), I don't actively follow this list. Thanks. Marc __ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.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 ___ --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] Developing VoIP with Asterisk
Hi Group!Hi Wagner!Thanks for the interest. I'm from Colombia and I'm trying to develop VoIP as you know on *. So thanks again for the offering in Brazil, althought you can help me with some idea by this way. To make the call I'm using SJphone (softphones) to make the tests. I'm not using IP phones because we don't have a lot of investment as a said before.This is my [general]sip.conf format: I omitted other parts which were on comments because it is example from the web site [general]context=default;allowguest=no ;realm=mydomain.tld bindport=5060bindaddr=0.0.0.0srvlookup=yes ;domain=mydomain.tld ;** Cambio de lineasdisallow=all;allow=g729allow=gsmallow=ulawjitterbuffer=yesmaxjitterbuffer=800;allow=ilbc;musicclass=default;language=en;relaxdtmf=yesrtptimeout=60 ;rtpholdtimeout=300;trustrpid = no;sendrpid = yes;progressinband=never ;useragent=Asterisk PBX;promiscredir = no ;usereqphone = no ;*** Cambio de lineas DTMFMODE estaba en comentarios dtmfmode = rfc2833 ;compactheaders = yes ;sipdebug = yes ;subscribecontext = default ;notifyringing = yes ; Usuario 1 [usuario1] type=friendhost=dynamicdtmfmode=rfc2833username=usuario1secret=usuario1; Usuario 2 [usuario2]type=friendhost=dynamicdtmfmode=rfc2833 username=usuario2secret=usuario2Thanks again for the interest and if you have and idea I would apreciate a lot!Carlos Bernat2006/7/27, Wagner Nunes [EMAIL PROTECTED]: Hi Carlos!!!Let me ask one thing... ... r u brazilian???Becouse I work with * projects and if u r in brazil maybe i can help u. But about your problem, What are u using to call thru *? IP Phone, softphone? What is your sip.conf settings? Carlos Alberto Bernat Orozco [EMAIL PROTECTED] escreveu: Hi Group!Still I'm concern about my problem with echo on the voice and I want to ask some advice to developing VoIP. Maybe I'm very ambiciuos or maybe not because I want to give VoIP to near 500 users. We got an small ISP and we have the project to give telephony (for now) to our users between them. Our resources are limited and I have installed * as a hope to give this service to our users. We have a good network (with small problems) but I believed that is possible to give this service. Our HFC network is very well calibrated and works fine. The users have cable modems to connect to the internet and we give private adresses to some users. I'm searching for someone who has the same problem in the past with similar things, to know how solve it and if is possible to give VoIP calls with a server with a public address and the softphones (for the costs) with extensions registered on our * box. I configured * four months ago and between two extensions and works very well and but later I did the same test on this week and unfortunaly the voice goes out with echo. So I have the feeling that maybe there's something wrong with the codecs and wich codecs do I need to give the service. Thanks for any help you can give meCarlos Bernat___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Você quer respostas para suas perguntas? Ou você sabe muito e quer compartilhar seu conhecimento? Experimente o Yahoo! Respostas! ___ --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] Developing VoIP with Asterisk
Hi Group!Still I'm concern about my problem with echo on the voice and I want to ask some advice to developing VoIP. Maybe I'm very ambiciuos or maybe not because I want to give VoIP to near 500 users.We got an small ISP and we have the project to give telephony (for now) to our users between them. Our resources are limited and I have installed * as a hope to give this service to our users. We have a good network (with small problems) but I believed that is possible to give this service. Our HFC network is very well calibrated and works fine. The users have cable modems to connect to the internet and we give private adresses to some users. I'm searching for someone who has the same problem in the past with similar things, to know how solve it and if is possible to give VoIP calls with a server with a public address and the softphones (for the costs) with extensions registered on our * box. I configured * four months ago and between two extensions and works very well and but later I did the same test on this week and unfortunaly the voice goes out with echo. So I have the feeling that maybe there's something wrong with the codecs and wich codecs do I need to give the service. Thanks for any help you can give meCarlos Bernat ___ --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] Developing VoIP with Asterisk (hardphones softphones)
Hi Group!Thanks Joshua Colp for your answer!!. I thinking (aprox. ) 50 simultaneous users up at the beginning. Some of our users, most, have private adresses, automatically assigned by dhcp. Some other, very few, has the public ones. I was thinking on hardphones because the idea for the user to pick up a real phone and make the call but our investment it's too short and I tried the SJphone for first time because only have to download. (is there a better?). But what you tell me it's right. I'll try to use hardphones to make some tests and I will give the results on this list if there is someone with similar troubles. I'm thinking to use some ATA's like cisco for testing. Any idea is wellcome!Carlos Bernat Hi Group!Greetings and salutations. Still I'm concern about my problem with echo on the voice and I want to ask some advice to developing VoIP. Maybe I'm very ambiciuos or maybe not because I want to give VoIP to near 500 users.The thing that you should be thinking about is how many simultaneous calls you want up. It's easy to have 500 accounts on a box. We got an small ISP and we have the project to give telephony (for now) to our users between them. Our resources are limited and I have installed * as a hope to give this service to our users. We have a good network (with small problems) but I believed that is possible to give this service. Our HFC network is very well calibrated and works fine. The users have cable modems to connect to the internet and we give private adresses to some users.This shouldn't be too bad. Do some users have public and some private? You may be able to get away with reinviting internally. This way Asterisk would not handle the audio. I'm searching for someone who has the same problem in the past with similar things, to know how solve it and if is possible to give VoIP calls with a server with a public address and the softphones (for the costs) with extensions registered on our * box. I configured * four months ago and between two extensions and works very well and but later I did the same test on this week and unfortunaly the voice goes out with echo. So I have the feeling that maybe there's something wrong with the codecs and wich codecs do I need to give the service.You should try with a hardphone so you can eliminate one of the variables in the equation. I have heard of problems previously where users were using softphones and they were introducing the echo. Switching to hardphones solved it and narrowed down the problem ;) Thanks for any help you can give meYou're welcome and hopefully some others can give some insight and maybe information on their own deployments similar to what you wish to do. Carlos Bernat Joshua ColpDigium ___ --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: Voice with echo
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