[asterisk-users] DECT with handover
Hello list, I'm looking for a solution where I need a DECT system with handover or repeaters for at least 10 simultaneous calls. The Snom m9 supports handover but I believe it's incompatible with asterisk since it requires to have the same SIP account associated with multiple IPs. I'm looking at Polycom's wireless servers but I couldn't figure out exactly how they work. Does the KIRK Wireless Server 300 support handover in a asterisk compatible way? Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] maximizing sound quality in 10.0
On Ter, 2011-12-27 at 13:34 -0600, Danny Nicholas wrote: Hi list, I have a set of 300 or so WAV files I was combining and playing using playback/background in 1.4.X. Now that I have moved on to the 10.0 set, I understand that I can replace my 8 Khz mono files with virtually unlimited Khz mono files (still no stereo, but a quantum leap forward). I’ve played with this and get good throughputs using SLIN44 formats on SIP. The 2 questions I have are: 1. Is Slin44 the format I should be settling on or has someone found a combination they find preferable? 2. While the SIP connections sound good, I still have to “talk” through OBI110 DADHI devices and other UUCM type connections – any pointers for juicing up the sound there? I'm not sure, but unless you're strictly talking through SIP _and_ using devices that support more than 8KHz, you won't take advantage of more than that. I believe that ISDN (BRI or PRI) and analogue lines as well as most phones use 8KHz. But, like I said, I'm not sure. Maybe someone can confirm that? I got curious myself. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use different local IP for each SIP trunk
Hello, Douglas Mortensen wrote: With that said, then it appears that the only way that I can have multiple trunks setup with them is to have asterisk use a different IP for all of the SIP RTP traffic for each given trunk. Essentially I would setup multiple IP addresses on my eth0 interface. Is there a way in asterisk that I could configure it to use one local IP for the source in all SIP/RTP traffic for 1 SIP trunk then a different local IP for the other SIP trunk? It's not an asterisk configuration but rather a interface configuration. I need something similar and I use 2 IPs on the same port. In debian, the configuration goes like this: auto eth0 iface eth0 inet static address ip1 netmask netmask1 network network1 broadcast broadcast1 gateway default_gateway auto eth0:0 iface eth0:0 inet static address ip2 netmask netmask2 up route add -net network2 netmask netmask2 gw gw2 And you can add more routes for other specific IPs/networks. José Pablo Méndez Soto wrote: May I ask why do you need different IP addresses to source calls? I mean, its not a common practice, would like to understand the idea behind it. In my case, the operator installed a gateway with a dedicated line and it's connected to the local network, but instead of being 192.168.0.0 it's on 10.0.0.0. So I use this 2 networks in the same NIC in the asterisk machine. Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.x segfaulting daily
Hello list,
An Asterisk installation that was doing fine suddenly stared segfaulting
a couple of times per day. I enabled all the logging and debugging to
try to find a pattern but there was too much information to see exactly
where it broke. So I enabled core dump and did backtraces and all of
them seem to break on ast_setstate, setting the state to AST_STATE_DOWN.
That's pretty much the only thing I can make of it, don't even know if
that's correct.
Does anyone have any ideas on why this is happening? The backtrace is
attached.
P.S.: I've switched the whole hardware already, including the BRI card
(B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and
mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1.
Best regards,
Paulo Santos
Core was generated by `/usr/sbin/asterisk'.
Program terminated with signal 11, Segmentation fault.
[New process 21726]
[New process 24376]
[New process 24375]
[New process 24374]
[New process 24371]
[New process 24344]
[New process 23560]
[New process 22868]
[New process 22329]
[New process 22327]
[New process 22325]
[New process 22324]
[New process 22323]
[New process 22322]
[New process 22321]
[New process 22320]
[New process 22319]
[New process 22318]
[New process 22317]
[New process 22316]
[New process 22315]
[New process 22259]
[New process 22208]
[New process 22203]
[New process 22185]
[New process 22184]
[New process 22160]
[New process 21515]
[New process 21725]
[New process 21687]
[New process 21686]
[New process 21685]
[New process 21681]
[New process 21659]
[New process 21658]
[New process 21648]
[New process 21647]
[New process 21609]
[New process 21594]
[New process 21542]
[New process 21540]
[New process 21516]
#0 0x080851ee in ast_setstate (chan=0xb3401c00, state=AST_STATE_DOWN) at
/usr/src/asterisk-1.4.42/include/asterisk/strings.h:37
37 return (!s || (*s == '\0'));
#0 0x080851ee in ast_setstate (chan=0xb3401c00, state=AST_STATE_DOWN) at
/usr/src/asterisk-1.4.42/include/asterisk/strings.h:37
name =
mISDN/4\000u\000ݴ��\177�\020\000@�\b(@�H0ݴf\211q�\020\000@�\b(@�\000\000@�Хe�\b(@�X\b@�h0ݴ\203\225a�P[
3] \000\000\000\000\000
#1 0xb561975d in release_chan (ch=0xb3400858, bc=0x88e8f5c) at
chan_misdn.c:3750
ast = (struct ast_channel *) 0xb3401c00
#2 0xb5622275 in cb_events (event=EVENT_CLEANUP, bc=0x88e8f5c, user_data=0x0)
at chan_misdn.c:4845
msn_valid = -1287644160
held_ch = value optimized out
ch = (struct chan_list *) 0xb3400858
__PRETTY_FUNCTION__ = cb_events
#3 0xb5632d9f in handle_cr (stack=0x88e82d8, frm=value optimized out) at
misdn/isdn_lib.c:1684
channel = 255
bc = (struct misdn_bchannel *) 0x88e8f5c
dummybc = {send_lock = 0xb67feff4, dummy = -1260570753, nt =
-1260572104, pri = -1234083825, port = -1260572068, b_stid = -1260571776,
layer_id = -1260570753, layer = -1234274741, need_disconnect = -1233129484,
need_release = -1260572068, need_release_complete = -1260571776,
dec = -1260571832, l3_id = -1234111388, pid = -1260572068, ces = -1251638304,
restart_channel = -1260570700, channel = -1260571776,
channel_preselected = 0, in_use = -1260571908, last_used = {tv_sec = 1023,
tv_usec = -72515583}, cw = -1260571776, addr = -1260571776,
bframe = 0xb4dd3380 handle_frm: frm-addr:42000303 frm-prim:3f182\n,
bframe_len = -1260571776, time_usec = -1260571729,
astbuf = 0xb4dd377f, misdnbuf = 0xb4dd3380, te_choose_channel = -1260570753,
early_bconnect = 0, dtmf = 0, send_dtmf = 0,
need_more_infos = 0, sending_complete = 0, nodsp = 1635021600, nojitter = 0,
dnumplan = NUMPLAN_UNKNOWN, rnumplan = 1308622848,
onumplan = NUMPLAN_UNKNOWN, cpnnumplan = NUMPLAN_UNINITIALIZED,
progress_coding = 824193585, progress_location = 942881334,
progress_indicator = 3617594, fac_in = {Function = Fac_GetSupportedServices,
u = {Listen = {NotificationMask = 21}, Suspend = {
CallIdentity = \025\000\000\000\000\000\000\000\000\000\000},
Resume = {
CallIdentity = \025\000\000\000\000\000\000\000\000\000\000},
CFActivate = {Handle = 21, Procedure = 0, BasicService = 0,
ServedUserNumber = \000\000\000\000Хe�\001\000\000,
ForwardedToNumber = @�\177�\000\000\000\000�wa�\0203ݴ,
ForwardedToSubaddress = \000\004\000\000�ze�7ݴ@�\177�}, CFDeactivate
= {Handle = 21, Procedure = 0, BasicService = 0,
ServedUserNumber = \000\000\000\000Хe�\001\000\000},
CFInterrogateParameters = {Handle = 21, Procedure = 0, BasicService = 0,
ServedUserNumber = \000\000\000\000Хe�\001\000\000},
CFInterrogateNumbers = {Handle = 21}, CDeflection = {
PresentationAllowed = 21, DeflectedToNumber =
\000\000\000\000\000\000\000\000\000\000Х,
DeflectedToSubaddress = e�\001\000\000\000@�\177�\000\000\000\000�w},
AOCDchu = {chargeNotAvailable = 21, freeOfCharge = 0,
recordedUnits = 0, typeOfChargingInfo = -1, billingId = 0}, AOCDcur =
{chargeNotAvailable
Re: [asterisk-users] Asterisk 1.4.x segfaulting daily
Hello, Thank you all for the replies. Steve Davies wrote: If I was guessing, I'd say that the channel structure that is being modified by the ast_setstate() call is incomplete, and contains some garbage pointers. If I was guessing further, I'd say that the callerID pointers are the most likely candidate - Does the issue happen when a caller-id withheld call is hung-up? hung-up before being answered perhaps? It was an outgoing call that tried to call through the port 2, then 1 and finally 3. The third port has a quite different debug output than the other 2. Maybe it's a problem on that connection, appears to be common on all segfaults. Apparently that third port is something of a strange group of BRI lines between that one and the line on the second port, but behaves differently. I'll try to find out more about it. Patrick Lists wrote: If the suggestion from Steve Davies doesn't work out for you then my suggestion would be to try out the latest DAHDI libpri with the latest Asterisk 1.8 because those versions have built-in support for the 4x BRI HFC chipset which can be found on the Digium b410p and the Openvox B400P. This way you no longer need mISDN V1 and have recent versions with tons of bugs fixed. Unfortunately I can't do that, at least not now. I will, however, migrate it eventually to either mISDN v2 or Dahdi, depending on the state of Dahdi then. P.S.: Attached the log. Best regards, Paulo Santos [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 333232837-5062-310@192.168.0.8 Their Tag 1036797295 Our tag: as5b7769e2 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 1693981358-5068-505@192.168.0.7 Their Tag 692402733 Our tag: as170cc25e [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 1394539361-5064-828@192.168.0.7 Their Tag 1627163612 Our tag: as5f15bf50 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 1708030692-5060-122@192.168.0.8 Their Tag 52015999 Our tag: as24b80c2d [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on RTP to Off [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on UDPTL to Off [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Allocating new SIP dialog for 1547819775-5062-295@192.168.0.7 - INVITE (With RTP) [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 0.0.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 192.168.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 10.0.0.0 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on RTP to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on UDPTL to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = Found Their Call ID: 1547819775-5062-295@192.168.0.7 Their Tag 2074339809 Our tag: as2515e4b3 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Received ACK (6) - Command in SIP ACK [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Stopping retransmission on '1547819775-5062-295@192.168.0.7' of Response 2940: Match Found [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = Found Their Call ID: 1547819775-5062-295@192.168.0.7 Their Tag 2074339809 Our tag: as2515e4b3 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 0.0.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 192.168.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 10.0.0.0 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on RTP to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on UDPTL to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP o=11 8002 8000 IN IP4 192.168.0.7... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.7... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:4 G723/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK. [Dec 12 16:38:36] DEBUG[22160
Re: [asterisk-users] ISDN - Busy signal on 3rd call
Paulo Santos wrote: Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] -- channel:0 mode:TE cause:0 ocause:0 rad: cad: The request from the telephone company's switch seems correct, a SETUP message (if 08 is Q.931, 05 is SETUP). 02 ff 03 08 01 04 05 a1 04 03 80 90 a3 18 01 80 6c 0b 01 83 39 31 36 33 39 31 37 34 32 70 03 c1 38 34 I've changed misdn.conf so it sends a release cause as 17 (user busy), but I get the same behaviour - cause:0 ocause:0. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo Santos misdn.conf: http://pastebin.com/FmgECqkU misdn debug: http://pastebin.com/Tg6wPKBD [1] http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html Ok, I've encountered a similar issue on a different installation but instead of being PTP it's PTMP. Plus, it's a setup with 2 BRI lines with call forwarding between them - main number of BRI1 forwards to secondary number of BRI2 when busy/unavailable and vice-versa. I've called the phone company and confirmed that call waiting is disabled, yet I get a message in misdn debug saying: P[ 2] -- Call Waiting on PMP sending RELEASE_COMPLETE I don't know if this is the actual call waiting feature or if it is just an information of some kind. In the misdn debug I get this: http://pastebin.com/D7wv0qqm The P[ 2] is the port of the BRI line I called in the first place, then it is forwarded to P[ 1] where I get an error: P[ 1] Decoding FACILITY failed! (-1) And the same issue I said in the previews email: P[ 1] -- channel:0 mode:TE cause:0 ocause:0 rad: cad: I changed isdn_lib.c and now I'm sending ocause:17 (user busy). I've done this in the PTP line mentioned in the previews email as well. For the PTP line it appears to have worked, I have the regular busy signal. It worked only after the first time I tried to place a 3rd call. Now the 3rd call doesn't even reach Asterisk, which was what I wanted from the phone company in the first place. On the PTMP line it didn't work, I still don't get the busy signal. Maybe cause 17 isn't the right one? And what can be that FACILITY mentioned in the debug? Thanks in advance. Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phones don't stop ringing
Hello list, I'm having some issues with some phones that don't stop ringing after the call is answered somewhere else. Basically, a call comes, enters a queue and all the phones in the queue ring. The problem is that when the call is answered, some phones don't stop ringing. I don't know if it is a configuration file, but I don't think so. queues.conf, sip.conf and extensions.conf: http://pastebin.com/8TTHpk4Z I've also captured a moment when this occurred: http://b.imagehost.org/0630/sip_flow.png The green one is the one that didn't stop ringing. The phone sends the first 180 Ringing _after_ the call is answered. This can be a network issue or a buggy firmware on the phones, but either way, shouldn't Asterisk send a CANCEL to an INVITE even if the phone didn't send 180 Ringing? Thanks in advance. Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Module reload
Hello, Flavio Miranda wrote: Every time I reload my asterisk it fall down and the following message appear on log: parse error: No category context for line 7 of /etc/asterisk/chan_dahdi.conf If I comment that line, it change to other line. There are some thing wrong with my dahdi? No, but there's probably something wrong with your chan_dahdi.conf. Maybe in the previews line. It would help if you could show us your chan_dahdi.conf Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN - Busy signal on 3rd call
Hello, Gopalakrishnan A.N wrote: I am also facing the call disconnection if there is a third call. I tried disable call waiting in the BRI router, but now it has been reduced, it means call disconnection is not permanent but seems to be occasion, let say per day two times there is a call disconnection. In the call disconnections after disabling call waiting, do you still get the following error as well? P[ 3] -- !! lib: No free channel! I've called the telephone company and they told me they had already disabled call waiting and answering machine, but because they pretty much have no idea what they're talking about I'll call them again and confirm those features are actually disabled. Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Module reload
Hello, Flavio Miranda wrote: Asterisk:/var/log/asterisk# pico /etc/asterisk/chan_dahdi.conf [...] You're missing the context [channels] at the start. Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN - Busy signal on 3rd call
I'm resending this email to the list, apparently the first one didn't go through. If it did, I apologize for the re-post. Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] -- channel:0 mode:TE cause:0 ocause:0 rad: cad: The request from the telephone company's switch seems correct, a SETUP message (if 08 is Q.931, 05 is SETUP). 02 ff 03 08 01 04 05 a1 04 03 80 90 a3 18 01 80 6c 0b 01 83 39 31 36 33 39 31 37 34 32 70 03 c1 38 34 I've changed misdn.conf so it sends a release cause as 17 (user busy), but I get the same behaviour - cause:0 ocause:0. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo Santos misdn.conf: http://pastebin.com/FmgECqkU misdn debug: http://pastebin.com/Tg6wPKBD [1] http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN - Busy signal on 3rd call
Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] -- channel:0 mode:TE cause:0 ocause:0 rad: cad: The request from the telephone company's switch seems correct, a SETUP message (if 08 is Q.931, 05 is SETUP). 02 ff 03 08 01 04 05 a1 04 03 80 90 a3 18 01 80 6c 0b 01 83 39 31 36 33 39 31 37 34 32 70 03 c1 38 34 I've changed misdn.conf so it sends a release cause as 17 (user busy), but I get the same behaviour - cause:0 ocause:0. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo Santos misdn.conf: http://pastebin.com/FmgECqkU misdn debug: http://pastebin.com/Tg6wPKBD [1] http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BRI line issue on third call
Hello list, I've been having a problem for some time now that I can't figure out how to solve it. On a PTP BRI ISDN line, if I have both channels in use and I place a third call from the outside, I'm not getting a busy tone like I should. Instead I get a congestion tone, as if the line was not working/didn't exist. Currently I'm using mISDN 1_1_9.1 and Asterisk 1.4.35 with an OpenVox B400P and I've tried all possible combinations of PTP/PTMP and with or without the termination jumper on the card. I've also tried to make it work since mISDN 1_1_8 and Asterisk 1.4.18. This is the message I get in Asterisk when the third call tries to come in: P[ 4] channel with stid:10020400 in use! P[ 4] channel with stid:10010400 in use! P[ 4] There is no free channel on port (4) P[ 4] -- !! lib: No free channel! P[ 4] I SEND:RELEASE_COMPLETE oad: dad: pid:0 P[ 4] -- channel:0 mode:TE cause:0 ocause:0 rad: cad: P[ 4] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 4] -- we have already send Release_complete asterisk/extensions.conf (relevant part): http://pastebin.com/a9nihVKt mISDN.conf: http://pastebin.com/6FPbcqc7 asterisk/misdn.conf: http://pastebin.com/9TGmGFFR Does anyone have any idea what can be causing this? Thanks in advance, Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't place 2nd call to provider
Hello list, I'm having problems placing the 2nd call via my provider. The first call goes through and I can talk normally, but when I place the second call, it doesn't go through and the first call is disconnected. The connection is 20mbps downstream and 1mbps upstream, so bandwidth is not an issue. I have another Asterisk running with the same configurations on another place and with the same provider and I don't have this issue. What can be the problem? Best regards, Paulo Santos ___ -- 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] Sound quality issue
Greetings everyone, I've been having some strange issues with my Asterisk box and some snom phones. In some cases, when I talk, the sound in the other end is cut off, I stop earing the background noise - looks like a walkie-talkie. I've tried this between phones in the same network and in all but one this happens. The one where it doesn't happen is the one connected directly to the router. I've tried different codecs with and without transcoding, including g722, and, on that phone, it all goes well. Could this have anything to do with the network? The main issue is that the router doesn't have QoS/ToS/whatever for me to test it. Plus, the phones are 3 switches away from the other phones. Thanks in advance. Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sound through NAT issue
Hello everyone, I'm having a hard time configuring my router to forward asterisk traffic correctly. I have the following ports being forwarded to asterisk: 5060, 1-2 Now, I can register the accounts when outside the network and I can call every extension that is inside the network. The problem is that I can't ear anything nor can the phones inside the network phone the outside phone. Is there any port I'm forgetting to forward? Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as a gateway
Greeting everyone, I'm trying to connect an old PBX to a Asterisk box with a 4 BRI card. The idea is for the PBX to follow asterisk's dialplan rules such as calling through VoIP when possible, ISDN when needed, etc, and all incoming calls being redirected to the PBX. The odd part is that incoming calls work perfectly, while when I make a call from a phone connected to the PBX through ISDN, I can hear the other party but they can't hear me and when the call is made through VoIP, I can't ear the ringing nor the other party (neither can the other party ear me), but the call is placed. I'm using alaw codec on every call. Does anyone have any idea what this problem could be? Thanks in advance, Best regards, Paulo Santos ___ -- 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] Asterisk on AVR32
Greetings, I'm sorry I've been taking so long to reply, but I've been swamped and didn't have the time to try to compile it. First of all, thank you all for the help. Kyle Kienapfel wrote: why is CROSS_ARCH=Linux? is this something the AVR32 distro is doing, or something you did? it should be something line avr or avr32 I pretty much reproduced all the variables AVR32's Buildroot was setting and applied it to ./configure on Asterisk. Tzafrir Cohen wrote: This error is not from Asterisk's configure. It is from 'make' running menuselect's configure. I meant that it is from 'make' with a regular ./configure, one with no parameters at all. While this is an ugly workaround, if you can't work with menuselect, try using my dummy-select. See the relevant parts of: http://git.tzafrir.org.il/?p=asterisk-tools.git;a=blob;f=git-asterisk-gui-howto Once I have a bit more time I'll try it out. Indeed my main problem seems to be menuselect. Doug Bailey wrote: When you run configure, you need to spec the host parameter for the architecture and environment you will be running under. I tried host=avr32-linux still with no success. The same error occurs, cannot execute binary file when it's trying to compile menuselect. I have compiled Buildroot with everything I need for Asterisk to work, only Asterisk itself isn't working. Does this mean that when I run 'make' on Buildroot it pretty much just runs 'make' on Asterisk with all those variables set? Or does it do some more operations? Thanks everyone, Best regards, Paulo Santos ___ -- 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 on AVR32
Greetings everyone, I'm trying to compile asterisk for an AVR32 (Atmel NGW100). Buildroot for AVR32 already has the asterisk package, though it has bugs. Firstly it tries to apply a patch for 1.2 on a 1.6, but deleting the contents of the patch file did the trick. Now, the problem is making asterisk. The first error is because asterisk needed to be ./configure:ed. Trying to just do ./configure, make gives an error [1]. Trying to do ./configure with the same args as make plus --host it can't even configure [2] I don't know much about cross-compiling, or even regular compiling for that matter. Does any one have any idea on how to do this? Thanks in advance, Best regards, Paulo Santos [1] menuselect/menuselect --check-deps menuselect.makeopts /bin/bash: menuselect/menuselect: cannot execute binary file make[1]: *** [menuselect.makeopts] Error 126 make[1]: Leaving directory `/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6' make: *** [/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6/asterisk] Error 2 [2] configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking build system type... i686-pc-linux-gnu checking host system type... Invalid configuration `CROSS_ARCH=Linux': machine `CROSS_ARCH=Linux' not recognized configure: error: /bin/bash ./config.sub CROSS_ARCH=Linux failed ___ -- 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] Digits timeout (ISDN)
Greetings everyone, I'm having some issues connecting a Asterisk box to a old ISDN PBX. Everything works fine but the undetermined digits rules. For instance, if I have _00X. and I start dialing for instance 0035..., Asterisk just get the 4 first numbers and starts dialing 0035. I've tried adding: stop_tone_after_first_digit = yes append_digits2exten = yes Although I believe these are the defaults. I was wondering if something like Set(TIMEOUT(digit)=5) would work in this situation. Has anyone had a similar problem? If so, how did you work around it? Thanks in advance, Best regards, Paulo Santos ___ -- 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] Digits timeout (ISDN)
Paulo Santos wrote: I was wondering if something like Set(TIMEOUT(digit)=5) would work in this situation. Found out Waitfordigits is needed in these situations. To make it available on asterisk I just downloaded app_bundle [1] and a simple make make install did the trick. Best regards, Paulo Santos [1] http://www.beronet.com/downloads/apps/app_bundle.tar.gz ___ -- 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] A400P + Intel D201GLY2(A) motherboard?
Gilles wrote: Hello I'd like to build myself an Asterisk server for SOHO use. Intel's D201GLY2 motherboard (http://tinyurl.com/ddarzp) looks like a very good deal, but I'm concerned about two things: 1. Will an A400P (from OpenVox, but supposed to be Digium-compatible http://tinyurl.com/ck6nfu) fit with a PCI riser, or will the CPU heatsink/fan be in the way? I downloaded the PDF from OpenVox's site, but didn't find the size of each PCI card The card goes the other way, it doesn't go on top of the board. Well, at least there are risers going away from the board, I don't know if there are any going on top of it. 2. Performance, especially if there's the need for software echo cancelling I did some tests on it, not many. Without going higher than 2.0 load average I managed to do 10 calls per second, lasting 5 seconds each. During those 5 seconds, 2 sound files were played (sln). MySQL CDR was enabled, so that's also 10 DB writes/second. I don't know exactly what board it was, but the processor was a Atom =2,2GHz. It had fan. Two cards were used at the same time, one B400P and one A800, both Openvox. ___ -- 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] A400P + Intel D201GLY2(A) motherboard?
Paulo Santos wrote: I managed to do 10 calls per second, lasting 5 seconds each. 10 or 5, I can't remember... ___ -- 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/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai
Gavin Henry wrote: Hi All, We've got msidn configured: Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - childcnt: 2 I don't know if it depends on the card, but in my case I need to set the termination jumper on TE mode for lines from PSTN. Mind to check the TE/NT jumper as well. te_ptmp=1 (...) [isdn] ports=1 context=from-pstn msns=* Here you have set PTMP, usually used with DID, and then you have MSN, usually used with PTP (at least here is how my telephone company use it). Find out if you use PTP or PTMP. Regards, Paulo Santos -- HTML e-mail is evil. Go plain text. ___ -- 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] CDR - Asterisk-Stat and PHP5
Tiago Durante wrote: Hi all, I don't know if its the right place to ask, but... Does any one have the asterisk-stat-v2 running with PHP5? Tks! # php --version PHP 5.2.0-8+etch13 (cli) (built: Oct 2 2008 08:26:18) Copyright (c) 1997-2006 The PHP Group Zend Engine v2.2.0, Copyright (c) 1998-2006 Zend Technologies Working for me. Don't forget you need php5-gd for the graphics to show. -- HTML e-mail is evil. Go plain text. ___ -- 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] CDR - Asterisk-Stat and PHP5
Marco Signorini wrote: Hi Tiago. I've it working on PHP 5.2.6 but only after having modified the php.ini default configuration keys: zend.ze1_compatibility_mode = Off short_open_tag = Off Though my zend.ze1_compatibility_mode is set to Off, short_open_tag is set to On and it is working. Those are my defaults, at least I never changed them. Installed with apt-get on Debian 4.0, PHP version 5.2.0-8+etch13. -- HTML e-mail is evil. Go plain text. ___ -- 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] DTMF tones mid conversation
Andrew Thomas wrote: I seem to have a problem of intermittent DTMF tones being played during a conversation. I'm having the same problem, but in my case, it's every 1 minute and at the start of the call. I wonder if it has anything to do with echo cancellation. I've only noticed when using a Zap channel, but I'll run some more tests. asterisk 1.4.17 / addons 1.4.7 / zaptel 1.4.12.1 / mISDN 1.1.8 Paulo Santos ___ -- 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
