[asterisk-users] SMS problems.
Hello, I tried to send sms for local extensions and i observed that file is created but sms isn't delivered yet. Can someone help me with this thing? rr:/var/spool/asterisk/sms/mttx # cat ../../outgoing/smsq.mttx.0.1322430026-20217.1 Channel: Local/1010 Callerid: SMS 1010 Application: SMS Data: 0,s MaxRetries: 0 RetryTime: 30 WaitTime: 10 rr:/var/spool/asterisk/sms/mttx # ls -la total 4 drwxr-xr-x 2 root root 88 Nov 27 23:40 . drwxr-xr-x 6 root root 144 Nov 27 23:40 .. -rw-r--r-- 1 root root 21 Nov 27 23:40 0.1322430026-20217 rr:/var/spool/asterisk/sms/mttx # cat 0.1322430026-20217 oa=1010 ud=TEST SMS. Thank you. P.S. I use smsq and asterisk 1.8.8.0. -- _ -- 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] many sip dialog/ opened channels.
Hello, I'm using asterisk with 84 extensions (aprox 45 always connected). When i look to the opened channels i sow many channels opened without reason even i don't have any active calls. Is there someone else that en-counted the same problem? Is there any fix to this bug? I have the following settings: Global Settings: UDP Bindaddress:[::]:5060 ** Additional Info: [::] may include IPv4 in addition to IPv6, if such a feature is enabled in the OS. TCP SIP Bindaddress:[::]:5060 TLS SIP Bindaddress:Disabled Videosupport: Yes Textsupport:Yes Ignore SDP sess. ver.: Yes AutoCreate Peer:No Match Auth Username:No Allow unknown access: No Allow subscriptions:Yes Allow overlap dialing: Yes Allow promisc. redir: No Enable call counters: Yes SIP domain support: Yes Realm. auth:No Our auth realm sip.someprovider.info Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: Yes Always auth rejects:Yes Direct RTP setup: No User Agent: asterisk SDP Session Name: Asterisk PBX 1.8.7.0 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify:No Legacy userfield parse: No Caller ID: asterisk From: Domain: sip.someprovider.info Record SIP history: On Call Events:On Auth. Failure Events: Off T.38 support: Yes T.38 EC mode: FEC T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 5000 ms Q.850 Reason header:No Store SIP_CAUSE:No Network QoS Settings: --- IP ToS SIP: CS3 IP ToS RTP audio: EF IP ToS RTP video: AF41 IP ToS RTP text:AF41 802.1p CoS SIP: 3 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 4 802.1p CoS RTP text:3 Jitterbuffer enabled: Yes Jitterbuffer forced:No Jitterbuffer max size: 300 Jitterbuffer resync:1000 Jitterbuffer impl: fixed Jitterbuffer log: No Network Settings: --- SIP address remapping: Disabled, no localnet list Externhost: none Externaddr: (null) Externrefresh: 5 Global Signalling Settings: --- Codecs: 0xe (gsm|ulaw|alaw) Codec Order:ulaw:20,alaw:20,gsm:20 Relax DTMF: No RFC2833 Compensation: Yes Symmetric RTP: No Compact SIP headers:No RTP Keepalive: 0 (Disabled) RTP Timeout:120 RTP Hold Timeout: 600 MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: No Pedantic SIP support: No Reg. min duration 30 secs Reg. max duration: 80 secs Reg. default duration: 1800 secs Outbound reg. timeout: 30 secs Outbound reg. attempts: 5 Notify ringing state: Yes Include CID: Yes Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: Yes Outb. proxy:not set Session Timers: Refuse Session Refresher: uas Session Expires:1800 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B:32000 No premature media: Yes Max forwards: 70 Default Settings: - Allowed transports: UDP Outbound transport: UDP Context:default Force rport:No DTMF: rfc2833 Qualify:500 Use ClientCode: No Progress inband:Yes Language: en MOH Interpret: default MOH Suggest:default Voice Mail Extension: voicemail and the opened channels: rr-de*CLI sip show channels Peer User/ANR Call ID Format Hold Last MessageExpiry Peer 6.6.13.17 (None) 000750d5-411d00 0x0 (nothing)No Rx: REGISTER guest 6.1.13.17 (None) 7de7064b-6f9f69 0x0 (nothing) No Rx: REGISTER guest 6.1.18.13 (None) 08a2e79c7f13b73 0x0 (nothing)No Rx: REGISTER guest 1.2.12.23 (None) 000dbcd9-39db00 0x0 (nothing)No Rx: REGISTER guest 8.6.13.17 (None) 000750d5-411d00 0x0 (nothing)No Rx: REGISTER guest 8.1.13.17 (None) ca30cc15-d93e4d 0x0 (nothing)No Rx: REGISTER guest 6.1.12.17 (None) 226b901d-4bff19 0x0 (nothing) No Rx: REGISTER guest 9.1.12.20 (None) 2474013819@192_ 0x0 (nothing) No Rx: REGISTER guest 2.1.14.10 (None) d1bb5072-b6ebcd 0x0 (nothing)No Rx:
[asterisk-users] Failure to write to tcp/tls socket
Hello, I have a strange situation with my asterisk 1.8.7.0 version. I compiled as usual everything seems to be ok but from time to time when i look on my console i get the following error message: [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket [Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread: Failure to write to tcp/tls socket -- SIP/3004-0001 is ringing [Oct 11 14:44:53] WARNING[20511]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x8a86560 (len 563) to 192.168.1.120:5080 returned -2: Success [Oct 11 14:44:53] WARNING[20511]: translate.c:162 framein: no samples for ulawtolin [Oct 11 14:45:03] WARNING[20511]: chan_sip.c:18940 function_sipchaninfo_read: This function can only be used on SIP channels. [Oct 11 14:45:03] WARNING[17330]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 144b033829324b1742f4f4f257df4...@intervoip.com for seqno 268 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 16400ms with no response Can somebody tell me how can i fix that error? I also look at opened channels and i saw with command sip show channels manny opened channels aprox 4000. That thing i suppose will rise load on processor and memory of my computer. Did someone else en-counted the same situation? Thank you for support. -- _ -- 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] single registration per user
Hello, I use asterisk 1.8.6.0 and I have aprox 100 extensions. I want to lock every extension to a single registration per device. Many of users tried to log on my asterisk from 2, 3 devices and I want allow only one. Is there any solution for fix this? Thank you. -- _ -- 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] Sip re-register / delay problem.
Hello, Can someone help me with some tips on this? many thanks On Wed, Sep 14, 2011 at 5:03 PM, Catalin S. jonsonpla...@gmail.com wrote: Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good response time to be check from time to time. defaultexpiry = 900 defaultexpirey = 900 maxexpiry = 300 maxexpirey = 300 minexpiry = 60 registerattempts = 5 registertimeout = 5 rtpholdtimeout = 900 rtptimeout = 60 jbmaxsize = 60 jbresyncthreshold = 200 qualify = yes qualify = 600 qualifyfreq = 60 Thank you. P.S. If you consider that i use too much options you can tell me what to drop. I use asterisk 1.8.6.0. -- _ -- 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] single registration per user
Hello Eric, Is about outgoing calls from multiple devices with the same username at aprox same time. The overwritten is for incomming calls. I want to prevent using the same account in multiple devices at same time. The solution with IP will not apply because users may be behind nat or will change everytime multiple access points. Do you have any other clues? Thank you for answers, Best regards. On Sun, Sep 18, 2011 at 8:37 PM, Eric Wieling ewiel...@nyigc.com wrote: Asterisk only allows one device per peer to register. If a 2nd device registers, the first registration is overwritten. You can use permit/deny to limit which IPs a device can register from. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S. Sent: Sunday, September 18, 2011 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] single registration per user Hello, I use asterisk 1.8.6.0 and I have aprox 100 extensions. I want to lock every extension to a single registration per device. Many of users tried to log on my asterisk from 2, 3 devices and I want allow only one. Is there any solution for fix this? Thank you. -- _ -- 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 -- _ -- 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] Sip re-register / delay problem.
Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good response time to be check from time to time. defaultexpiry = 900 defaultexpirey = 900 maxexpiry = 300 maxexpirey = 300 minexpiry = 60 registerattempts = 5 registertimeout = 5 rtpholdtimeout = 900 rtptimeout = 60 jbmaxsize = 60 jbresyncthreshold = 200 qualify = yes qualify = 600 qualifyfreq = 60 Thank you. P.S. If you consider that i use too much options you can tell me what to drop. I use asterisk 1.8.6.0. -- _ -- 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] Variables error in 1.8.6.0.
Hello Leandro,
Can you tell me a short example about how can i use what you gave me for
instance suppose i want to use { txjitter, DBL, { .d8 =
stats.txjitter, }, }, how can i set it in CDR variable like mine:
exten = h,n,set(CDR(ljitt)=${CHANNEL(rtpqos,audio,local_jitter)})
Thank you.
On Mon, Sep 5, 2011 at 10:58 PM, Leandro Dardini ldard...@gmail.com wrote:
2011/9/5 Catalin S. jonsonpla...@gmail.com
Hello,
I have a problem with some variables in 1.8.6.0. I set on extension the
following lines:
exten = h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio,
local_lostpackets)}) ; lost packets by local end **
exten = h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio,
remote_lostpackets)}) ; lost packets by remote end
exten = h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos, audio,
local_jitter)}) ; the Same for jitter
Theoretically this should throw these variables in a table in MySQL but
these values cannot be readed. I think it's a different syntax in
1.8.
I gave this error:
- Executing [h @ macro-special1: 11] Set (SIP/1010-0002, CDR
(LLP) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio,
remote_lostpackets' to CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
remote_lostpackets'
- Executing [h @ macro-special1: 12] Set (SIP/1010-0002, CDR
(PCR) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, local_jitter' to
CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
local_jitter'
- Executing [h @ macro-special1: 13] Set (SIP/1010-0002, CDR
(ljitt) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, remote_jitter'
to CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
remote_jitter'
Any idea how I can fix?
Best regards,
Jonson.
--
It is really simple, a patch of few months ago renamed the vars, but
forget to update the documentation. You have to use the source for finding
the new variable names. I paste here the part of the code for your easy
viewing...
{ txcount, INT, { .i4 =
stats.txcount, }, },
{ rxcount, INT, { .i4 =
stats.rxcount, }, },
{ txjitter, DBL, { .d8 =
stats.txjitter, }, },
{ rxjitter, DBL, { .d8 =
stats.rxjitter, }, },
{ remote_maxjitter, DBL, { .d8 =
stats.remote_maxjitter, }, },
{ remote_minjitter, DBL, { .d8 =
stats.remote_minjitter, }, },
{ remote_normdevjitter, DBL, { .d8 =
stats.remote_normdevjitter, }, },
{ remote_stdevjitter,DBL, { .d8 =
stats.remote_stdevjitter, }, },
{ local_maxjitter, DBL, { .d8 =
stats.local_maxjitter, }, },
{ local_minjitter, DBL, { .d8 =
stats.local_minjitter, }, },
{ local_normdevjitter, DBL, { .d8 =
stats.local_normdevjitter, }, },
{ local_stdevjitter, DBL, { .d8 =
stats.local_stdevjitter, }, },
{ txploss, INT, { .i4 =
stats.txploss, }, },
{ rxploss, INT, { .i4 =
stats.rxploss, }, },
{ remote_maxrxploss, DBL, { .d8 =
stats.remote_maxrxploss, }, },
{ remote_minrxploss, DBL, { .d8 =
stats.remote_minrxploss, }, },
{ remote_normdevrxploss, DBL, { .d8 =
stats.remote_normdevrxploss, }, },
{ remote_stdevrxploss, DBL, { .d8 =
stats.remote_stdevrxploss, }, },
{ local_maxrxploss, DBL, { .d8 =
stats.local_maxrxploss, }, },
{ local_minrxploss, DBL, { .d8 =
stats.local_minrxploss, }, },
{ local_normdevrxploss, DBL, { .d8 =
stats.local_normdevrxploss, }, },
{ local_stdevrxploss,DBL, { .d8 =
stats.local_stdevrxploss, }, },
{ rtt, DBL, { .d8 =
stats.rtt, }, },
{ maxrtt,DBL, { .d8 =
stats.maxrtt
[asterisk-users] Variables error in 1.8.6.0.
Hello,
I have a problem with some variables in 1.8.6.0. I set on extension the
following lines:
exten = h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio,
local_lostpackets)}) ; lost packets by local end **
exten = h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio,
remote_lostpackets)}) ; lost packets by remote end
exten = h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos, audio, local_jitter)})
; the Same for jitter
Theoretically this should throw these variables in a table in MySQL but
these values cannot be readed. I think it's a different syntax in 1.8.
I gave this error:
- Executing [h @ macro-special1: 11] Set (SIP/1010-0002, CDR
(LLP) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio,
remote_lostpackets' to CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
remote_lostpackets'
- Executing [h @ macro-special1: 12] Set (SIP/1010-0002, CDR
(PCR) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, local_jitter' to
CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
local_jitter'
- Executing [h @ macro-special1: 13] Set (SIP/1010-0002, CDR
(ljitt) =) in new stack
[September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221
sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, remote_jitter'
to CHANNEL
[September 5 22:39:33] WARNING [14432]: func_channel.c: 393
func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio,
remote_jitter'
Any idea how I can fix?
Best regards,
Jonson.
--
_
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[asterisk-users] Bind SIP over TCP port in asterisk 1.4.42.
Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Thank you very much. Jonson. -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
hello, I tried still not working. :( something is wrong. On Thu, Aug 25, 2011 at 4:37 PM, Skyler skchopper...@gmail.com wrote: Hi, On Thu, 2011-08-25 at 16:26 +0300, Catalin S. wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Not 100% with 1.4 but with 1.6 you don't need to set tcpbindaddr because tcpenable will listen on same IP as udp. No transport either I believe. If you want, set udpbindaddr and tcp will listen on this IP too. tcpenable=yes is all you should need. S. -- _ -- 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 -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
Hello, I tried but still not works. Can you make some test at your side? Something is wrong. Thank you. On Thu, Aug 25, 2011 at 4:35 PM, Andrew Latham lath...@gmail.com wrote: On Thu, Aug 25, 2011 at 9:26 AM, Catalin S. jonsonpla...@gmail.com wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Thank you very much. Jonson. -- I looked TCP + Transport are listed in http://svn.asterisk.org/svn/asterisk/branches/1.4/channels/chan_sip.c but not in http://svn.asterisk.org/svn/asterisk/branches/1.4/configs/sip.conf.sample try transport=TCP Beware, some systems use SIP(not encrypted) over TCP on port 5061, which is not really wrong, just not what the standards say. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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 -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
Hello Paul, I choose 1.4.42 version because of iksemel for gtalk module. I need gtalk feature for my asterisk and when i upgrade it at 1.6 this module doesn't work. Can you tell me if is some trick to make 1.4.42 to work with tcp option? Maybe some patches... etc. Thank you. On Thu, Aug 25, 2011 at 5:25 PM, Paul Belanger pabelan...@digium.comwrote: On 11-08-25 09:26 AM, Catalin S. wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not see tcp port raised up. Did somebody had the same problem and had some solutions? Asterisk 1.4 does not have support SIP over TCP. It was added in Asterisk 1.6.0. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Bind SIP over TCP port in asterisk 1.4.42.
Thank you Paul for answers. Please tell me if i upgrade to 1.8 is gtalk module still working? On Thu, Aug 25, 2011 at 5:42 PM, Paul Belanger pabelan...@digium.comwrote: On 11-08-25 10:34 AM, Catalin S. wrote: Hello Paul, I choose 1.4.42 version because of iksemel for gtalk module. I need gtalk feature for my asterisk and when i upgrade it at 1.6 this module doesn't work. Can you tell me if is some trick to make 1.4.42 to work with tcp option? Maybe some patches... etc. Well, both 1.4 and 1.6 branches are unsupported so you should move to asterisk 1.8 and test. There is no 'trick' for adding TCP over SIP into Asterisk 1.4, that is not realistic. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Prepay Limited Calls.
Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really want to install another software to make this or modify all my settup. I'm wonder if someone is using something simple to limmit calls. Anyway if someone is using some other programs/software/scripts and another settup/method please let me know how is yours. I want to check few methods to realize that limmit. Thank you for help guys. -- _ -- 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] Optimize peers registration under jitter/delay.
Hello, I want to optimize my registrations and calls of peers to my asterisk with the following options in sip.conf: ---///--- qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using operator Internet with delay/jitter conditions? I chooses values above after many tests but still have some problems: - from time to time peers have lagged connections... maximum time to re register is 10 seconds... can i minimize that or refresh without unregister and re register? i want that users to be as much as he can online even in delay/jitter conditions. Of course if is response time too much like over 1000, 2000 ms i prefer to re register if it can. - is any connection between these timers for keepalive connections , re register etc... and choppy sounds/sometimes interrupted/nosy, in an active call? If yes how can i optimize both things: to hav' a good sound and to keepalive connections for peers. Thank you very much for support... please feel free to ask me any question or misunderstanding of this mail, and I'll email you with more detail. -- _ -- 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] Optimize peers registration under jitter/delay.
did you also hav qualify and qualifyfreq? Thank you for reply, On Mon, Jul 26, 2010 at 1:55 PM, Faisal Hanif fai...@vopium.com wrote: We are having good results with maxexp 120 minexp 90 defexp 100 qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using operator Internet with delay/jitter conditions? I chooses values above after many tests but still have some problems: - from time to time peers have lagged connections... maximum time to re register is 10 seconds... can i minimize that or refresh without unregister and re register? i want that users to be as much as he can online even in delay/jitter conditions. Of course if is response time too much like over 1000, 2000 ms i prefer to re register if it can. - is any connection between these timers for keepalive connections , re register etc... and choppy sounds/sometimes interrupted/nosy, in an active call? If yes how can i optimize both things: to hav' a good sound and to keepalive connections for peers. Thank you very much for support... please feel free to ask me any question or misunderstanding of this mail, and I'll email you with more detail. -- _ -- 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 -- _ -- 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] Callin Numbers.
Hello, I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? Thank you very much, Jonson. ___ -- 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] Callin Numbers.
On Wed, Jul 22, 2009 at 2:41 PM, Geoff Lanege...@gjctech.co.uk wrote: On Wednesday, July 22, 2009, Catalin S. wrote: I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? There are several providers who offer UK or US regional geographical numbers for little or no cost if you only use them inbound. For example, I have UK geographicals from Sipgate (http://www.sipgate.co.uk/user/index.php) and VoipCheap (http://www.voipcheap.com/en/index.html - *not voipcheap.co.uk*). The latter, I had to install their client to a Windows host and inspect the configuration to obtain the info necessary to connect my Asterisk server. However, those are only examples and there are a lot more to be found if you look around. HTH, -- Geoff ___ -- 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
Re: [asterisk-users] Callin Numbers.
Hello sorry for earlier message, I push send before write something. Anyway I tried that sites and also lowratevoip.com. All gives me the follwing message: Sorry – at this moment there are no VoIP-In numbers available for your country (yet). We will inform you as soon as there are (new) numbers available for your region. Click to go back. Do you have some tested sites please? Thank you. On Wed, Jul 22, 2009 at 2:41 PM, Geoff Lanege...@gjctech.co.uk wrote: On Wednesday, July 22, 2009, Catalin S. wrote: I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? There are several providers who offer UK or US regional geographical numbers for little or no cost if you only use them inbound. For example, I have UK geographicals from Sipgate (http://www.sipgate.co.uk/user/index.php) and VoipCheap (http://www.voipcheap.com/en/index.html - *not voipcheap.co.uk*). The latter, I had to install their client to a Windows host and inspect the configuration to obtain the info necessary to connect my Asterisk server. However, those are only examples and there are a lot more to be found if you look around. HTH, -- Geoff ___ -- 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] Double dial.
Hello, I have a strange situation with an SPA3102 FXO/FXS device. I'm in situation that when i receive a call from PBX line I must forward the calls to 2 VoIP numbers. Right now i have the following settings: (S0:1...@gw1). I want to forward at 1020 too. I tested (S0:1010|1...@gw1) and doesn't work. Did you have any other ideea? Thank you. ___ -- 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] Special Dialplan
Hello ppl,
I want to make a special dial plan for routing calls to a peer which
has an pin protection.
Normally if you want to call through that peer you must first enter
pin for example 1234#
and after that you hear the tone from line and after that you can dial
desired numbers.
I tried something like that, but doesn't worked. Did somebody have some clues?
exten = 0X.,n(dial1),Dial(SIP/peer-account/1234#${0x},15,rt)
Thank you guys for any help. I appreciate.
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Re: [asterisk-users] Special Dialplan
Thank you for your answer Steve,
Well I want to do this automatically... I mean if I want to route
0x through peer-account (which is pin protected), everything to be
automatically.
In fact i route through SIPURA SPA3102 Linksys fxo/fxs device , so my
device will answer and wait for my pin, if is ok wait for number to be
called.
Anyway did you know how can i send dtmf after is answered?
Thank you.
On Sat, May 9, 2009 at 11:45 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
On Sat, May 9, 2009 at 4:22 PM, Catalin S. jonsonpla...@gmail.com wrote:
Hello ppl,
I want to make a special dial plan for routing calls to a peer which
has an pin protection.
Normally if you want to call through that peer you must first enter
pin for example 1234#
and after that you hear the tone from line and after that you can dial
desired numbers.
I tried something like that, but doesn't worked. Did somebody have some
clues?
exten = 0X.,n(dial1),Dial(SIP/peer-account/1234#${0x},15,rt)
Thank you guys for any help. I appreciate.
Start with an Answer() and then lose the r from your dial string. That
should allow you to press the code in.
If you want to hard code it, use dial and then probably Wait() followed by
senddtmf.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] strange text message:)
I don't know what is MWI Message. All I know is that i can find these messages in my SMS inbox and has the sender voicem...@mydomain.xxx On 2/24/09, OCG Technical Support supp...@ocg.ca wrote: Are you sure this is not just a standard SIP MWI message? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S. Sent: February 23, 2009 8:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] strange text message:) is any chance to use this feature to send messages on this kind of phones? On Tue, Feb 24, 2009 at 1:39 AM, David fire ddf...@gmail.com wrote: you are getting the info about the voicemail becausethe soft on your phone support it. in sip.conf you can find some parameters to send that info. in other soft phones like x-lite you will have the same info. David 2009/2/23 Catalin S. jonsonpla...@gmail.com Hello guys, I recently observed that my asterisk sends me sms like messages on my phone (Nokia E71), I mean is SMS but is delivered some kind in-band though VoIP. Is strange because this messages contains informations about my voicemail and is sent by voicem...@mydomainxxx.com. I noticed that this messages appears every time when I logged in with my phone on my sip account. I'm interested about how can I send these messages with other information's or whatever I want to my terminals. Also I observed that works with Nokia E71 only. Maybe is because I updated some software on It , Not Firmware. Do you guys observed this too? Thank you for support. Catalin. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 ___ -- 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] DTMF Forwarking Problems.
Hello ppl, I have a problem with my asterisk when i want to call some destination through my peers and I must enter DTMF digits to select some extension/conference number or password to access some features.Every numbers is accepted but when i must press # key my asterisk interpret it like transfer options. I want to know how can i activate and deactivate transfer mode of # key on my desired peers. Thank you very much, Catalin. ___ -- 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] strange text message:)
Hello guys, I recently observed that my asterisk sends me sms like messages on my phone (Nokia E71), I mean is SMS but is delivered some kind in-band though VoIP. Is strange because this messages contains informations about my voicemail and is sent by voicem...@mydomainxxx.com. I noticed that this messages appears every time when I logged in with my phone on my sip account. I'm interested about how can I send these messages with other information's or whatever I want to my terminals. Also I observed that works with Nokia E71 only. Maybe is because I updated some software on It , Not Firmware. Do you guys observed this too? Thank you for support. Catalin. ___ -- 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] strange text message:)
is any chance to use this feature to send messages on this kind of phones? On Tue, Feb 24, 2009 at 1:39 AM, David fire ddf...@gmail.com wrote: you are getting the info about the voicemail becausethe soft on your phone support it. in sip.conf you can find some parameters to send that info. in other soft phones like x-lite you will have the same info. David 2009/2/23 Catalin S. jonsonpla...@gmail.com Hello guys, I recently observed that my asterisk sends me sms like messages on my phone (Nokia E71), I mean is SMS but is delivered some kind in-band though VoIP. Is strange because this messages contains informations about my voicemail and is sent by voicem...@mydomainxxx.com. I noticed that this messages appears every time when I logged in with my phone on my sip account. I'm interested about how can I send these messages with other information's or whatever I want to my terminals. Also I observed that works with Nokia E71 only. Maybe is because I updated some software on It , Not Firmware. Do you guys observed this too? Thank you for support. Catalin. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Cisco IP Phone 7940G.
Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get any luck here either. I look in the /var/log/messages and I observed that my phone request 4 different files that i don't have it in my tftp directory. Here's my tftp output session with my phone: Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to 192.168.1.3:52178 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml to 192.168.1.3:52180 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf to 192.168.1.3:52181 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to 192.168.1.3:52182 as you see my phone request 4 files that doesn't comes in archive P0S3-08-11-00.zip: SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf, SEPDefault.cnf... while my archive contents is the following: OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads, P0S3-08-11-00.sb2 3.) I want to make this phone to be SIP compatible. A friend of main gave me a .cnf file with an example of configuration for SIP. How may I rename this cnf file to make work with my phone. 4.) On the other side my phone doesn't have ringtone either. Any clue how may I put ringtones on it? I know is a lot of questions for you guys, but I browse on cisco.com web site and google for hours and I don't get it any clue to make work this phone in any way. Thank you for help. Jonson. ___ -- 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 IP Phone 7940G.
I understand, but i cannot load the new firmware... is any well know method? On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov abalas...@evaristesys.com wrote: This phone is currently running the SCCP (Skinny) image. Before you will get anywhere you need to load the SIP firmware image onto it. The SEP* configuration files are for SCCP. After doing that, the phone will start requesting the correct files. You may need to upgrade through various SIP images cumulatively. On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S. jonsonpla...@gmail.com wrote: Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get any luck here either. I look in the /var/log/messages and I observed that my phone request 4 different files that i don't have it in my tftp directory. Here's my tftp output session with my phone: Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to 192.168.1.3:52178 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml to 192.168.1.3:52180 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf to 192.168.1.3:52181 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to 192.168.1.3:52182 as you see my phone request 4 files that doesn't comes in archive P0S3-08-11-00.zip: SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf, SEPDefault.cnf... while my archive contents is the following: OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads, P0S3-08-11-00.sb2 3.) I want to make this phone to be SIP compatible. A friend of main gave me a .cnf file with an example of configuration for SIP. How may I rename this cnf file to make work with my phone. 4.) On the other side my phone doesn't have ringtone either. Any clue how may I put ringtones on it? I know is a lot of questions for you guys, but I browse on cisco.com web site and google for hours and I don't get it any clue to make work this phone in any way. Thank you for help. Jonson. ___ -- 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 -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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
Re: [asterisk-users] Cisco IP Phone 7940G.
hey finally i did it. I upgraded the firmware to the latest sip firmware and now i have the another problem. The requested files are the following: ---///--- Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf.xml to 192.168.1.3:51252 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIP00141CAA4B4C.cnf to 192.168.1.3:51253 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIPDefault.cnf to 192.168.1.3:51254 ---///--- I made my own sip configuration in SIP00141CAA4B4C.cnf where 00141CAA4B4C is the mac address of phone, but i don't know what to write in CTLSEP00141CAA4B4C.tlv, SEP00141CAA4B4C.cnf.xml and SIPDefault.cnf. On display of the screen of phone all I have is Tftp file missing... probably it expect all these files. Anyway, Ronny I can give you my archive with what i had in my tftp and i succeeded to update firmware. Just tell me if you want to send on your personal e-mail these files. Thank you guys for your help and interest. On 2/13/09, k4...@bellsouth.net k4...@bellsouth.net wrote: I'm trying to do the same and have read the mentioned sites. The one item I can't seem to get past is a working TFTP server. What is the easiest method to get one running and what packages in Linux or Windows work best? Thanks for putting up with a Linux newbie. Ronny -- Original message from Alex Balashov abalas...@evaristesys.com: -- Have a look at: http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080 094584.shtml#topic2 On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S. wrote: I understand, but i cannot load the new firmware... is any well know method? On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov wrote: This phone is currently running the SCCP (Skinny) image. Before you will get anywhere you need to load the SIP firmware image onto it. The SEP* configuration files are for SCCP. After doing that, the phone will start requesting the correct files. You may need to upgrade through various SIP images cumulatively. On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S. wrote: Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get any luck here either. I look in the /var/log/messages and I observed that my phone request 4 different files that i don't have it in my tftp directory. Here's my tftp output session with my phone: Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to 192.168.1.3:52178 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml to 192.168.1.3:52180 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf to 192.168.1.3:52181 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to 192.168.1.3:52182 as you see my phone request 4 files that doesn't comes in archive P0S3-08-11-00.zip: SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf, SEPDefault.cnf... while my archive contents is the following: OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads, P0S3-08-11-00.sb2 3.) I want to make this phone to be SIP compatible. A friend of main gave me a .cnf file with an example of configuration for SIP. How may I rename this cnf file to make work with my phone. 4.) On the other side my phone doesn't have ringtone either. Any clue how may I put ringtones on it? I know is a lot of questions for you guys, but I browse on cisco.com web site and google for hours and I don't get it any clue to make work this phone in any way. Thank you for help. Jonson. ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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
Re: [asterisk-users] Cisco IP Phone 7940G.
i finally did it... It works excellent. Thank you guys for help. On Fri, Feb 13, 2009 at 9:44 PM, David Gibbons d...@videon-central.com wrote: snip On a similar subject, I have been able to get a 7961 to switch to a SIP firmware, has anyone had any luck with this? /snip Yes, I have several 7961s and 7971s running SIP, same firmware generation as the 41s --Dave ___ -- 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] Loose connection with MySql.
Hello, I configured asterisk to use mysql for CDR. Well when i check from time to time I realize that asterisk loose connection with mysql (i use phpmyadmin and i watch the processes). Can anybody tell me how can i solve that problem? I want to have all cdr statistics logged in mysql, is very important for billing. Thank you for support. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] Loose connection with MySql.
Hello guys, thank you for all your answers. I'll will check and i keep you informed of what's happening next. Note that mysql and asterisk is on the same machine so is not a problem of connectivity or mysql machine to be down. On Tue, Jun 24, 2008 at 3:22 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 24 June 2008 04:43:19 Al Baker wrote: errr -you mean Asterisk doesn't ALWAYS check this and reconnect with the database ?!? WTF Since the CDRs are the literal Cash and Life Blood of many application why the heck would it NOT do this as part of its minimal basic operation ??? If it Doesn't do this for CDRs does it NOT do it for RealTime ?? If not, one could it up,screwed,blued and tatoed Is this functionality or lack there of documented anyplace ??? You might want to check your facts before launching into a diatribe. Both the MySQL backend driver for CDR as well as the MySQL backend driver for Realtime reconnect if possible during a query. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] application sendtext
Hello did you find something? I want to do the same thing. I have asterisk and nokia e51 phone.. Also i tried several models. On 5/23/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi, I want to send some text to the phone such that the phone can display the text on its display. I have tried to use SendText but it doesn't work. Does the phone need to support when asterisk issues the SendText application? ango ___ -- 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] Sending texts questions.
Hello, i have installed the latest asterisk software and I user soft phones and hard phones (generally Nokia E-Series with sip and wifi enabled functions). I want to know how may i send in band messages to my clients. Simple text messages on their devices/software - clients. Thank you for any ideas. Jonson. ___ -- 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] Problems with calls in asterisk.
Hello, i recently installed last version of asterisk (Asterisk 1.4.18.1 built by root @ h-gw on a i586 running Linux on 2008-03-23 00:26:44 UTC) and everything is ok but when i call an extension i cannot hear anything. I don't get any visible error on sip debug... i changed the codecs... everything is the same... Can someone help me with that? Thank you. Jonson. ___ -- 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] Stange pause between extensions commands.
Hello and thank you for reply... I tried with Playback() and is the same
effect. Is curious because sometime there's no pause other time is a long
pause.
Anybody have other idea?
Thank you.
On 12/14/07, Atis Lezdins [EMAIL PROTECTED] wrote:
On 12/14/07, Catalin S. [EMAIL PROTECTED] wrote:
Hello,
i have a simple but annoying problem. I have the following entry in
/etc/asterisk/externsions.conf file:
---Cut Here---
exten = 10100,1,Wait(4)
exten = 10100,2,Playback(transfer,noanswer)
exten = 10100,3,Dial(${PHONE30},30,t)
exten = 10100,4,Background(extension)
exten = 10100,5,Background(is-curntly-unavail)
Why do you have Background() here? I think it should be Playback()
Regards,
Atis
exten = 10100,6,Voicemail()
exten = 10100,7,PlayBack(vm-goodbye)
exten = 10100,8,Hangup
---And Here---
Normally when i call that extension if the user is online will ring if
not,
will play: Extension is currently unavailable and immediately should
go to
voicemail and after voicemail will play: Good bye and hangup. But
after
plain Extension is currently unavailable is a long period of silence
and
finally will go to voicemail. On my asterisk i have the following output
during this call:
---Cut Here---
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/10100-082244c0,
SIP/1010|20)
in new stack
[Dec 14 14:37:05] WARNING[20355]: app_dial.c:1131 dial_exec_full:
Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2]
BackGround(SIP/10100-082244c0, extension) in new stack
-- SIP/10100-082244c0 Playing 'extension' (language 'en')
-- Executing [EMAIL PROTECTED]:3]
BackGround(SIP/10100-082244c0, is-curntly-unavail) in
new stack
-- SIP/10100-082244c0 Playing 'is-curntly-unavail' (language
'en')
-- Executing [EMAIL PROTECTED]:4] VoiceMail(SIP/10100-082244c0,
10100)
in new stack
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP
still
has peer field or pending or callno (flags = 16, peer = 0x8189c00 callno
=
0)
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP
still
has peer field or pending or callno (flags = 16, peer = 0x82084d0 callno
=
0)
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP
still
has peer field or pending or callno (flags = 16, peer = 0x81e6d98 callno
=
0)
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP
still
has peer field or pending or callno (flags = 16, peer = 0x81daf00 callno
=
0)
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP
still
has peer field or pending or callno (flags = 16, peer = 0x81e4cc8 callno
=
0)
[Dec 14 14:37:12] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten o
[Dec 14 14:37:17] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten o
[Dec 14 14:37:22] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten o
[Dec 14 14:37:27] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten o
[Dec 14 14:37:32] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten a
[Dec 14 14:37:37] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten a
[Dec 14 14:37:42] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten a
[Dec 14 14:37:47] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten a
-- SIP/10100-082244c0 Playing 'vm-intro' (language 'en')
== Spawn extension (default, 10100, 4) exited non-zero on
'SIP/10100-082244c0'
---And Here---
Can anyone help me with this? I want immediately voicemail answer...
maybe
these error is the cause... I saw that in this pause the asterisk tried
to
contact this extension through my external peers (genetically named
sip.xxx.com)... Thank you...
___
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--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
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___
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[asterisk-users] Stange pause between extensions commands.
Hello,
i have a simple but annoying problem. I have the following entry in
/etc/asterisk/externsions.conf file:
---Cut Here---
exten = 10100,1,Wait(4)
exten = 10100,2,Playback(transfer,noanswer)
exten = 10100,3,Dial(${PHONE30},30,t)
exten = 10100,4,Background(extension)
exten = 10100,5,Background(is-curntly-unavail)
exten = 10100,6,Voicemail()
exten = 10100,7,PlayBack(vm-goodbye)
exten = 10100,8,Hangup
---And Here---
Normally when i call that extension if the user is online will ring if not,
will play: Extension is currently unavailable and immediately should go to
voicemail and after voicemail will play: Good bye and hangup. But after
plain Extension is currently unavailable is a long period of silence and
finally will go to voicemail. On my asterisk i have the following output
during this call:
---Cut Here---
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/10100-082244c0, SIP/1010|20) in
new stack
[Dec 14 14:37:05] WARNING[20355]: app_dial.c:1131 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] BackGround(SIP/10100-082244c0,
extension) in new stack
-- SIP/10100-082244c0 Playing 'extension' (language 'en')
-- Executing [EMAIL PROTECTED]:3] BackGround(SIP/10100-082244c0,
is-curntly-unavail) in new stack
-- SIP/10100-082244c0 Playing 'is-curntly-unavail' (language 'en')
-- Executing [EMAIL PROTECTED]:4] VoiceMail(SIP/10100-082244c0, 10100)
in new stack
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still has
peer field or pending or callno (flags = 16, peer = 0x8189c00 callno = 0)
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still has
peer field or pending or callno (flags = 16, peer = 0x82084d0 callno = 0)
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still has
peer field or pending or callno (flags = 16, peer = 0x81e6d98 callno = 0)
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still has
peer field or pending or callno (flags = 16, peer = 0x81daf00 callno = 0)
[Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still has
peer field or pending or callno (flags = 16, peer = 0x81e4cc8 callno = 0)
[Dec 14 14:37:12] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten o
[Dec 14 14:37:17] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten o
[Dec 14 14:37:22] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten o
[Dec 14 14:37:27] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten o
[Dec 14 14:37:32] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten a
[Dec 14 14:37:37] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten a
[Dec 14 14:37:42] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten a
[Dec 14 14:37:47] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
waiting for xxx:[EMAIL PROTECTED] exten a
-- SIP/10100-082244c0 Playing 'vm-intro' (language 'en')
== Spawn extension (default, 10100, 4) exited non-zero on
'SIP/10100-082244c0'
---And Here---
Can anyone help me with this? I want immediately voicemail answer... maybe
these error is the cause... I saw that in this pause the asterisk tried to
contact this extension through my external peers (genetically named
sip.xxx.com)... Thank you...
___
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[asterisk-users] Call back or some voicemail notifing.
Hello PPL, someone have any idea for notifying users that they have voicemail waiting when they will register after weren't being registered on asterisk? I need this for nokia terminal e series users. I studied sms service but seems to be only for PSTN lines. I comes with idea to receive a call from asterisk and notified that you have a voicemail. Thank you. ___ --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 back voicemail.
Hello ppl, is any set of configuration for asterisk that could put asterisk to call users when they come back online in case they have any voicemail? I think is a good modality to inform users that they have a voicemail and listen to it. Thank you for you support. ___ --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] Introducing myself
Welcome Andres, we will keep in touch:) On 8/16/07, Andres Jimenez [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, all First post to a new (for me!)list. Netiquette as a must. My name is Andres Jimenez and I am an spaniard working as System Administrator in Dublin (Ireland). I just started working with Asterisk, but thanks to all the available documentation the community has created I an being able to get over any problem in my VoIP setup. I want to thank you all for your previous and future help. - -- Andres Jimenez -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGxCoP8SZxpGYWwpYRAjaVAKDdJ/8H0O7Cx/hLmwDI/7XQARag+gCg1uGN f1bzk8UGM97F+4Elciip7og= =j6ET -END PGP SIGNATURE- ___ --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
