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Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In the asterisk console i am seeing "Everyone is busy/congested at this time". In AMP - Outbound dialing i hv configured a rou...

...trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe <201> This is rather odd, as if I create a nearly identical call file in /var/spool/asterisk/outgoing (below) the receiving phone rings correctly. Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priorit...

...1 on Zap/55-1 Mar 26 09:51:18 DEBUG[3896] chan_sip.c: Auto destroying call 'a5306fd4-bcea3062-caa16c03@192.168.3.2' Mar 26 09:51:18 DEBUG[4885] chan_zap.c: Enabled echo cancellation on channel 55 Mar 26 09:51:18 VERBOSE[4885] logger.c: -- Executing Goto("Zap/55-1", "to-sip|201|1") in new stack Mar 26 09:51:18 VERBOSE[4885] logger.c: -- Goto (to-sip,201,1) Mar 26 09:51:18 VERBOSE[4885] logger.c: -- Executing Dial("Zap/55-1", "SIP/201@192.168.2.13|120") in new stack Mar 26 09:51:18 DEBUG[4885] chan_sip.c: Outgoing Call for 201 Mar 26 09:51:18 VERB...

I'm setting up a new install of samba 2.2.7a on FreeBSD 4.7 Here's a little run-through of what I have done. I added a machine account in /etc/group as follows: machines:*:201 I have added all of my machine accounts in to /etc/passwd as follows: sclepy$:*:201:201::0:0:Machine account for Sclepy:/dev/null:false lexus$:*:202:201::0:0:Machine account for Lexus:/dev/null:false kellogg$:*:203:201::0:0:Machine account for Kellogg:/dev/null:false april$:*...

Hello All, I have one X100P card for inbound calls. I use two Broadvoice SIP accounts for all my outbound calls. I'm unable to place calls using BV. Inbound BV calls are ok. Verbosity is at least 3 -- Executing Macro("SIP/201-365c", "dialout-default|XXXXXXX") in new stack -- Executing GotoIf("SIP/201-365c", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/201-365c", ""The Shaws"?6") in new stack -- Executing...

...ndcron' logged off from 127.0.0.1 [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] <--- SIP read from UDP:192.168.0.28:5060 ---> REGISTER sip:192.168.0.99 SIP/2.0 Call-ID: d7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0 CSeq: 4 REGISTER From: "201" <sip:201 at 192.168.0.99>;tag=5fbdd638 To: "201" <sip:201 at 192.168.0.99> Max-Forwards: 70 User-Agent: Jitsi2.6.5390Mac OS X Expires: 600 Contact: "201" <sip:201 at 192.168.0.28:5060;transport=udp;registering_acc=192_168_0_99>;exp...

Hi all, Maybe I'm being thick here, but I've had a look through the mailing list and the Wiki, and I can't seem to see details of anybody else with this problem.... With the following line: exten => s,1,HasNewVoicemail(201) I am getting the following error: -- Executing HasNewVoicemail("SIP/201-2f1e", "201") in new stack Sep 2 12:41:09 NOTICE[819221]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 201 at /var/spool/asterisk/voicemail/default/201/(null) does not exist Sep 2 12...

...ernal network are routed correctly. The problem is with the packets that originate from the router. For some reason they are routed through the default interface. Does anybody know why does that happen? # ip rule ls 0: from all lookup local 200: from 192.168.1.0/24 to a.b.c.0/24 lookup 202 201: from e.f.g.0/24 lookup 201 201: from 192.168.1.0/24 to 62.44.96.0/19 lookup 201 201: from 127.0.0.0/8 to 62.44.96.0 lookup 201 202: from a.b.c.0/24 lookup 202 32766: from all lookup main 32767: from all lookup default # ip route ls e.f.g.0/24 dev eth3 scope link metric 1 a.b.c.0/2...

I am running Asterisk@Home Voicemail works fine but does not email out the voicemail attachments. Any suggestion? ----------------------------------- Voicemail.conf [general] #include vm_general.inc #include vm_email.inc [default] 201 => {password},Jeff G Laptop,jrglass@columbus.rr.com,,attach=yes --------------------------------------------------------------------- Sip.Conf [201] username=201 type=friend secret={ACCOUT PASSWORD} qualify=no port=5060 nat=yes mailbox=201 host=dynamic dtmfmode=rfc2833 context=from-internal can...

I have a dataset that I'm trying to rearrange for a repeated measures analysis: It looks like: patient basefev1 fev11h fev12h fev13h fev14h fev15h fev16h fev17h fev18h drug 201 2.46 2.68 2.76 2.50 2.30 2.14 2.40 2.33 2.20 a 202 3.50 3.95 3.65 2.93 2.53 3.04 3.37 3.14 2.62 a 203 1.96 2.28 2.34 2.29 2.43 2.06 2.18 2.28 2.29 a 204 3.44 4.08 3.87 3.79 3.30 3.80 3.24 2.98 2.91 a And I...

...quot;ss7/08053004990|60", action=0) at pbx.c:544 e = (struct ast_exten *) 0x9e15c28 sw = (struct ast_switch *) 0x0 data = 0x0 foundcontext = 0xa281970 "default" newstack = 1 res = 0 status = 5 incstack = {0xa32e21 "\201??!\f", 0xf6a2e099 "X\215e?[^_??U\211?\213U\f?", 0xf3879570 "\001", 0xf6be7574 "0\017|", 0xf3c00010 "", 0xaf4ff4 "<M?", 0xf3c00010 "", 0x15 "", 0xf46a60ec "?`j?!.?", 0xa334ba "e\203=\f", 0xf3c00010 &q...

...Host Dyn Nat ACL Mask Port Status 204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN 203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms) 202/202 (Unspecified) D 255.255.255.255 0 UNKNOWN 201/201 192.167.125.12 D 255.255.255.255 5060 OK (3 ms) moloch*CLI> as you can see, 201 and 203 are on-line but, if i call from 203 to 201, i immediately go to voicemail instead of doing call to 201. Here's the SIP call flow: moloch*CLI> Sip read: INVITE si...

...y running on centrala (pid = 28749) -- Remote UNIX connection Verbosity was 3 and is now 11 centrala*CLI> MANAGER LOGIN MD5 127.0.0.1, admin, amp111 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin' logged on from 127.0.0.1 -- Executing Queue("SIP/201-ec33", "prodaja") in new stack -- Started music on hold, class 'default', on SIP/201-ec33 -- Stopped music on hold on SIP/201-ec33 -- Playing 'queue-youarenext' (language 'si') -- Told SIP/201-ec33 in prodaja their queue position (which was 1)...

Hello all, I've installed AAH 1.1 on my VIA C3 powered mini PC. I've made the necessary changes to the * makefile, so the compilation went well. The first thing I did was configuring two extensions from AMP, namely 200 and 201. Then I installed X-lite on two PC's and configured them with one of the extensions: System settings - SIP proxy - Default: Username: 200 Authorisation user: 200 Password: **** Domain/Realm: babbelbox SIP Proxy: babbelbox Babbelbox is the hostname of my * server and DNS is working properly. No...

.../pabx-canall/${EXTEN},60,tT) > > exten => _2XX,1,Answer() exten => _2XX,n,MixMonitor(${CALLERID(num)}-$ > {STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav) > exten => _2XX,n,Dial(SIP/${EXTEN},60,tT) > > The scenario is as following: > > 1) 201 asks operator for an external call, hangs up. The audio file is > stored correctly. From the CLI: > > [Jan 8 16:20:19] -- Executing [200 at default:1] Answer("SIP/ > 201-081d8740", "") in new stack > [Jan 8 16:20:19] -- Executing [200 at default:2] MixMonitor(...

Hi, Realtime SIP and Extensions are working fine but facing some problems with Voicemail. Added an entry to extconfig.conf voicemail => mysql,asterisk,voicemail_users Created the corresponding table and an entry for mailbox 201. This is also reflected in the CLI as shown below. CLI> realtime load voicemail mailbox 201 Column Name Column Value -------------------- -------------------- uniqueid 1 customer_id 201 mailbox 201...

...rs Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No No 202 duccio from-internal No No 201 fabrizio from-internal No No moloch*CLI> it's ok. So i use kphone to connect top my asterisk server. KPhone say that i'm on-line so i'll check "sip show registry" and it's empty: moloch*CLI> sip show reg...

Hello, I am having trouble creating a (1000,3,201) array in R from a data set created within the same script. My problem is that I want to take a matrix that is 1000x603 (called "landmat") and separate this into 201 separate (1000x3) matrices. My problem is when I try to do particular combinations of the columns. I want to take the fi...

...er. The docs on the Dial command dont mention any special characters. I'd say someone must have come up against this before for other services too that require * or # characters. Any help would be great! Thanks as always, Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: derek@rivertower.ie Web: www.rivertowerhosting....

...not correcly registering with asterisk. The phone boots DHCP gets an address, loads the SIP software and sets there for me to dial. However, I get the INV when I dial. Any ideas on why the phone is displaying invalid and what to do about it??? Thanks, jerry sip.conf ------------------------ [201] type=friend dtmfmode=rfc2833 username=201 secret=201 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid="Media Assistant" <201> [202] type=friend dtmfmode=rfc2833 username=202 secret=202 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip c...