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:
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
