[asterisk-users] Progress audio associated with 180 Ringing not passed to extension when using pjsip
I’m setting up a new PBX in the Google cloud running FreePBX 14.0.1.36 / Asterisk 14.7.5. Most calls are fine, but when calling an AT landline that is busy, ringback tone is heard instead of the expected busy signal. An example of a failing number is +1 408 269 1999 (a test number that is always busy). My production system running FreePBX 2.11.0.43 / Asterisk 11.4.0 using chan_sip and the same phone and trunk does not have this problem. Both extension and AnveoDirect trunk are using pjsip. I’m using Anveo’s ‘Smart Route Option’ which sends the call directly to AT (no intermediate tandem carrier). I don’t understand why, but AT’s response to the INVITE is 180 Ringing with associated SDP. RTP (containing audio for busy signal) starts coming in, but the 180 Ringing passed to the extension does not have associated SDP so the phone continues to play ringback. If I change the trunk to use chan_sip instead, the problem disappears. There is also no problem if I force Anveo to send the call to Verizon (which acts as a tandem carrier for this call) ; Verizon sends 183 Progress with the busy signal audio. Tests with Flowroute did show the trouble (I assume that they would send this call directly to AT). I’d like to migrate to pjsip – is there a trunk setting or manual config edit that works around this issue? Or is it somehow related to my build and does not affect other users? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] new install: no re-invite and unwanted transcoding
I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on internal calls, Asterisk only proxies audio when necessary, no unwanted transcoding. For initial testing, I've set up two Yealink T26P extensions and one Localphone trunk. Internal and external calls work, except for the problems above. The extensions are behind a NAT, but are set up with STUN, unique SIP and RTP ports, and proper forwarding. The router handles hairpin connections properly. When registered to the old system, calls between the test extensions re-invite correctly. On the new system, no re-invites are attempted and I see nothing logged to indicate why. Re-invite also fails on inbound and outbound trunk calls, and on trunk-to-trunk calls (tested by setting follow-me to an external number). The extensions are coded with: Asterisk Dial Options: r canreinvite: Yes nat: No - RFC3581 disallow: all allow: g722ulawalaw Recording Options (all): Never The trunk (both PEER and USER Details) has: canreinvite=yes In Advanced Settings - Device Settings I have: SIP canrenivite (directmedia): Yes In Asterisk SIP Settings I have: NAT: No IP Configuration: Public IP Codecs: ulaw, alaw Reinvite Behavior: Yes Other settings are defaults, except for a non-standard bindport. An entry from sip_additional.conf, as generated by FreePBX: [1001] deny=0.0.0.0/0.0.0.0 disallow=all secret=password dtmfmode=rfc2833 canreinvite=yes context=from-internal host=dynamic trustrpid=yes sendrpid=pai type=friend nat=no port=5060 qualify=yes qualifyfreq=60 transport=udp avpf=no icesupport=no encryption=no callgroup= pickupgroup= allow=g722 allow=ulaw allow=alaw dial=SIP/1001 mailbox=1001@device permit=0.0.0.0/0.0.0.0 callerid=John Doe 1001 callcounter=yes faxdetect=no cc_monitor_policy=generic The dial command produced by FreePBX also looks reasonable: -- Executing [s@macro-dial-one:43] Dial(SIP/1002-007e, SIP/1001,,rI) in new stack A second issue is that on outbound PSTN calls, Asterisk is accepting the phone's first-preference codec (g722), speaking ulaw on the trunk side and transcoding, resulting in degraded quality. Incoming calls escape this problem; Asterisk offers ulaw/g722/alaw, the phone accepts the first (ulaw) and no transcoding occurs. How can I tell Asterisk to prefer ulaw over g722, when it would otherwise need to transcode? (The transcoding issue also affects the old system, but I gave up debugging it and just disabled g722 on the phones.) Any advice will be gratefully appreciated. Thanks, Stewart -- _ -- 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] Problem getting TDM400P clone card to go off-hook and dial
I am having problems getting a Nicherons TDM400P wildcard clone to dial out.
Everything appears to be configured correctly, but although I see call
progress, it never seems to actually pick up the phone.
(The following is a test of 911 emergency, where I substitute 811 [repair
service] as the actual number dialed.)
*CLI
-- Executing [911@from-internal:1] Goto(SIP/101-,
nineoneone,s,1) in new stack
-- Goto (nineoneone,s,1)
-- Executing [s@nineoneone:1] Set(SIP/101-,
SET_EMERG_FLAG=0) in new stack
-- Executing [s@nineoneone:2] ChanIsAvail(SIP/101-, DAHDI/4)
in new stack
-- Hanging up on 'DAHDI/4-1'
-- Hungup 'DAHDI/4-1'
-- Executing [s@nineoneone:3] Set(SIP/101-,
GLOBAL(EMERGENCY)=1) in new stack
== Setting global variable 'EMERGENCY' to '1'
-- Executing [s@nineoneone:4] Set(SIP/101-,
SET_EMERG_FLAG=1) in new stack
-- Executing [s@nineoneone:5] Dial(SIP/101-, DAHDI/4/811) in
new stack
-- Called 4/811
-- DAHDI/4-1 is ringing
== Extension Changed 101[from-internal] new state Idle for Notify User 101
== Extension Changed 101[from-internal] new state Idle for Notify User 103
-- Hanging up on 'DAHDI/4-1'
-- Hungup 'DAHDI/4-1'
== Spawn extension (nineoneone, s, 5) exited non-zero on
'SIP/101-'
-- Executing [h@nineoneone:1] GotoIf(SIP/101-, 1?3) in new
stack
-- Goto (nineoneone,h,3)
-- Executing [h@nineoneone:3] Set(SIP/101-,
GLOBAL(EMERGENCY)=0) in new stack
== Setting global variable 'EMERGENCY' to '0'
*CLI
When DAHDI/4-1 is ringing appears I indeed hear ringing progress tones,
but they appear to be coming from Asterisk, as the card does not pick up the
phone at this point, or ever. I'm using jack #4 on the board, which is
supposedly an FXO port.
Here's the output from various relevant tools config files:
--
*CLI dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState
pseudodefault
default In Service
1from-pstn
default In Service
2from-pstn
default In Service
3from-internal
default In Service
4from-internal
default In Service
---
dahdi-channels.conf:
; Autogenerated by /usr/sbin/dahdi_genconf on Fri Apr 1 06:52:48 2011
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is
intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global
settings
;
; Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER)
;;; line=1 WCTDM/4/0 FXSKS (SWEC: MG2)
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 1
callerid=
group=
context=default
;;; line=2 WCTDM/4/1 FXSKS (SWEC: MG2)
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 2
callerid=
group=
context=default
;;; line=3 WCTDM/4/2 FXOKS (SWEC: MG2)
signalling=fxo_ks
callerid=Channel 3 4003
mailbox=4003
group=5
context=from-internal
channel = 3
callerid=
mailbox=
group=
context=default
;;; line=4 WCTDM/4/3 FXOKS (SWEC: MG2)
signalling=fxo_ks
callerid=Channel 4 4004
mailbox=4004
group=5
context=from-internal
channel = 4
callerid=
mailbox=
group=
context=default
; Span 2: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: HRtimer) 1
--
chan_dahdi.conf
...
...
[channels]
#include /etc/asterisk/dahdi-channels.conf
...
-
root@Trixie:/etc/asterisk# dahdi_hardware
pci::02:01.0 wctdm+ e159:0001 Wildcard TDM400P REV I
---
root@Trixie:~# lsdahdi
### Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER)
1 FXOFXSKS (In use) (SWEC: MG2)
2 FXOFXSKS (In use) (SWEC: MG2)
3 FXSFXOKS (In use) (SWEC: MG2)
4 FXSFXOKS (In use) (SWEC: MG2)
### Span 2: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: HRtimer) 1
root@Trixie:~#
-
extensions.conf (excerpt)
; Global variables
[globals]
; Stuff for 911
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/4
; Change this for production use:
;EMERGENCY_NUM=some_test_phone_number
EMERGENCY_NUM=811
;EMERGENCY_NUM=911
...
; Which trunk to use for any DAHDI (PSTN-'Hard Line'-AKA POTS) type stuff
POTSTRUNK=DAHDI/4
...
; Emergency -- DO NOT REMOVE!
exten = 911,1,Goto(nineoneone,s,1)
...
; EMERGENCY! See http://www.voip-info.org/wiki-Asterisk+tips+911 for
details.
[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,Set(GLOBAL(EMERGENCY)=1)
exten = s,n,Set(SET_EMERG_FLAG=1)
exten
[asterisk-users] Problem getting TDM400P clone card to go off-hook and dial
I am having problems getting a Nicherons TDM400P wildcard clone to dial out.
Everything appears to be configured correctly, but although I see call
progress, it never seems to actually pick up the phone.
(The following is a test of 911 emergency, where I substitute 811 [repair
service] as the actual number dialed.)
*CLI
-- Executing [911@from-internal:1] Goto(SIP/101-,
nineoneone,s,1) in new stack
-- Goto (nineoneone,s,1)
-- Executing [s@nineoneone:1] Set(SIP/101-,
SET_EMERG_FLAG=0) in new stack
-- Executing [s@nineoneone:2] ChanIsAvail(SIP/101-, DAHDI/4)
in new stack
-- Hanging up on 'DAHDI/4-1'
-- Hungup 'DAHDI/4-1'
-- Executing [s@nineoneone:3] Set(SIP/101-,
GLOBAL(EMERGENCY)=1) in new stack
== Setting global variable 'EMERGENCY' to '1'
-- Executing [s@nineoneone:4] Set(SIP/101-,
SET_EMERG_FLAG=1) in new stack
-- Executing [s@nineoneone:5] Dial(SIP/101-, DAHDI/4/811) in
new stack
-- Called 4/811
-- DAHDI/4-1 is ringing
== Extension Changed 101[from-internal] new state Idle for Notify User 101
== Extension Changed 101[from-internal] new state Idle for Notify User 103
-- Hanging up on 'DAHDI/4-1'
-- Hungup 'DAHDI/4-1'
== Spawn extension (nineoneone, s, 5) exited non-zero on
'SIP/101-'
-- Executing [h@nineoneone:1] GotoIf(SIP/101-, 1?3) in new
stack
-- Goto (nineoneone,h,3)
-- Executing [h@nineoneone:3] Set(SIP/101-,
GLOBAL(EMERGENCY)=0) in new stack
== Setting global variable 'EMERGENCY' to '0'
*CLI
When DAHDI/4-1 is ringing appears I indeed hear ringing progress tones,
but they appear to be coming from Asterisk, as the card does not pick up the
phone at this point, or ever. I'm using jack #4 on the board, which is
supposed an FXO port,
Here's the output from various relevant tools config files:
--
*CLI dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState
pseudodefault
default In Service
1from-pstn
default In Service
2from-pstn
default In Service
3from-internal
default In Service
4from-internal
default In Service
---
dahdi-channels.conf:
; Autogenerated by /usr/sbin/dahdi_genconf on Fri Apr 1 06:52:48 2011
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is
intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global
settings
;
; Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER)
;;; line=1 WCTDM/4/0 FXSKS (SWEC: MG2)
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 1
callerid=
group=
context=default
;;; line=2 WCTDM/4/1 FXSKS (SWEC: MG2)
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 2
callerid=
group=
context=default
;;; line=3 WCTDM/4/2 FXOKS (SWEC: MG2)
signalling=fxo_ks
callerid=Channel 3 4003
mailbox=4003
group=5
context=from-internal
channel = 3
callerid=
mailbox=
group=
context=default
;;; line=4 WCTDM/4/3 FXOKS (SWEC: MG2)
signalling=fxo_ks
callerid=Channel 4 4004
mailbox=4004
group=5
context=from-internal
channel = 4
callerid=
mailbox=
group=
context=default
; Span 2: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: HRtimer) 1
--
chan_dahdi.conf
...
...
[channels]
#include /etc/asterisk/dahdi-channels.conf
...
-
root@Trixie:/etc/asterisk# dahdi_hardware
pci::02:01.0 wctdm+ e159:0001 Wildcard TDM400P REV I
---
root@Trixie:~# lsdahdi
### Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER)
1 FXOFXSKS (In use) (SWEC: MG2)
2 FXOFXSKS (In use) (SWEC: MG2)
3 FXSFXOKS (In use) (SWEC: MG2)
4 FXSFXOKS (In use) (SWEC: MG2)
### Span 2: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: HRtimer) 1
root@Trixie:~#
-
extensions.conf (excerpt)
; Global variables
[globals]
; Stuff for 911
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/4
; Change this for production use:
;EMERGENCY_NUM=some_test_phone_number
EMERGENCY_NUM=811
;EMERGENCY_NUM=911
...
; Which trunk to use for any DAHDI (PSTN-'Hard Line'-AKA POTS) type stuff
POTSTRUNK=DAHDI/4
...
; Emergency -- DO NOT REMOVE!
exten = 911,1,Goto(nineoneone,s,1)
...
; EMERGENCY! See http://www.voip-info.org/wiki-Asterisk+tips+911 for
details.
[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,Set(GLOBAL(EMERGENCY)=1)
exten = s,n,Set(SET_EMERG_FLAG=1)
exten
[asterisk-users] PRI D-channel bouncing
ââ Wildcard TE121 Card 0 F10=Back -- Andrew Stewart astew...@notre1.com -- _ -- 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] PRI D-channel bouncing
That was it. Thanks!!! On Tue, Aug 10, 2010 at 9:14 AM, Michael L. Young myo...@acsacc.com wrote: -- Michael L. Young Administrative Claim Service, Inc. | IT Manager 600 Main Street, Suite 5, Winchester, MA 01890 www.acsacc.com Phone 781-721-1998 - Original Message - From: Andrew Stewart astew...@notre1.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 10, 2010 9:33:45 AM Subject: [asterisk-users] PRI D-channel bouncing I need some help getting a system running for one of my company's plants. I am running AsteriskNow 1.7 with Asterisk 1.6.2.10 and FreePBX 2.8.0.2. My D-Channel keeps bouncing. The telecom tech told me he thought that I might be using the wrong sync source, and I think I might have been, but I changed DAHDI system.conf to span=1,1,0,ESF,B8ZS (from span=1,0,0,ESF,B8ZS) and I am still having the same problem. (Although, the FreePBX DAHDI page only allows me to select 0 in the Sync/Clock Source field. 0 is the only option in the drop down.) * [r...@gch-asterisknow01 ~]# cat /etc/asterisk/chan_dahdi_groups.conf ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make ; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; ; [span_1] signalling=pri_net switchtype=national pridialplan=national prilocaldialplan=national group=0 context=from-pstn channel = 1-15 Is the PRI coming from the telephone carrier? If so, shouldn't the signalling be pri_cpe? Michael L. Young (elguero) -- _ -- 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 -- Andrew Stewart astew...@notre1.com (205) 585-2980 - cell -- _ -- 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 reply CALL-ID from ITSP has internal address in host part
We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b1d72f1328c...@%externip% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b1d72f1328c...@%internip%. I can not figure out where the ITSP is even getting my %INTERNIP% from, I don't see it in the packet anywhere. I have externip, localnet, and nat=yes all setup in my sip.conf. Any ideas of where to look for the source of this problem? -aws -- Andrew Stewart astew...@notre1.com (205) 585-2980 - cell ___ -- 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
Re: [asterisk-users] SIP reply CALL-ID from ITSP has internal address in host part
On Wed, Sep 9, 2009 at 8:59 AM, Alex Balashovabalas...@evaristesys.com wrote: Andrew Stewart wrote: We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b1d72f1328c...@%externip% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b1d72f1328c...@%internip%. I can not figure out where the ITSP is even getting my %INTERNIP% from, I don't see it in the packet anywhere. This doesn't seem quite right. If the 200 OK reply is truly for the INVITE (or whatever other transaction is initiated by your SIP packet), it *must* have the *same* Call-ID per the RFC, otherwise it's not a valid reply. The Call-ID is what's called a GUID (Globally Unique IDentifier). It is up to every SIP user agent to generate one, and the only requirement is that it be as unique as practical in time and SIP space. Many network elements like to tack on IP addresses in the GUID as a means of differentiating it further, though personally I think that's a bad idea. Would you mind pasting a capture of the transaction in question, from the vantage point of the outside interface of your Asterisk host? You can change the representations of the external IP to something else if you don't want to post it to a public list. Thanks, -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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 Wireshark export of two packets pasted below. I simply did a find/relace and put %EXTERNIP% in place of my actual public, PATed, IP address. That is only modification I did to these pcaps. No. TimeSourceDestination Protocol Info 1 0.00192.168.114.64209.62.1.2SIP Request: OPTIONS sip:sip.us1.voip.ms Frame 1 (544 bytes on wire, 544 bytes captured) Arrival Time: Sep 4, 2009 13:36:02.490711000 [Time delta from previous captured frame: 0.0 seconds] [Time delta from previous displayed frame: 0.0 seconds] [Time since reference or first frame: 0.0 seconds] Frame Number: 1 Frame Length: 544 bytes Capture Length: 544 bytes [Frame is marked: False] [Protocols in frame: eth:ip:udp:sip] [Coloring Rule Name: UDP] [Coloring Rule String: udp] Ethernet II, Src: Dell_95:35:26 (00:22:19:95:35:26), Dst: Cisco_7d:53:80 (00:0e:38:7d:53:80) Destination: Cisco_7d:53:80 (00:0e:38:7d:53:80) Address: Cisco_7d:53:80 (00:0e:38:7d:53:80) ...0 = IG bit: Individual address (unicast) ..0. = LG bit: Globally unique address (factory default) Source: Dell_95:35:26 (00:22:19:95:35:26) Address: Dell_95:35:26 (00:22:19:95:35:26) ...0 = IG bit: Individual address (unicast) ..0. = LG bit: Globally unique address (factory default) Type: IP (0x0800) Internet Protocol, Src: 192.168.114.64 (192.168.114.64), Dst: 209.62.1.2 (209.62.1.2) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 00.. = Differentiated Services Codepoint: Default (0x00) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Total Length: 530 Identification: 0x6abe (27326) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x08f4 [correct] [Good: True] [Bad : False] Source: 192.168.114.64 (192.168.114.64) Destination: 209.62.1.2 (209.62.1.2) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Source port: sip (5060) Destination port: sip (5060) Length: 510 Checksum: 0x0739 [validation disabled] [Good Checksum: False] [Bad Checksum: False] Session Initiation Protocol Request-Line: OPTIONS sip:sip.us1.voip.ms SIP/2.0 Method: OPTIONS Request-URI: sip:sip.us1.voip.ms Request-URI Host Part: sip.us1.voip.ms [Resent Packet: False] Message Header Via: SIP/2.0/UDP %EXTERNIP%:5060;branch=z9hG4bK69fa843c;rport Transport: UDP Sent-by Address: %EXTERNIP% Sent
Re: [asterisk-users] SIP reply CALL-ID from ITSP has internal address in host part
On Wed, Sep 9, 2009 at 10:45 AM, Andrew Stewart astew...@notre1.com wrote: On Wed, Sep 9, 2009 at 8:59 AM, Alex Balashovabalas...@evaristesys.com wrote: Andrew Stewart wrote: We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b1d72f1328c...@%externip% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b1d72f1328c...@%internip%. I can not figure out where the ITSP is even getting my %INTERNIP% from, I don't see it in the packet anywhere. This doesn't seem quite right. If the 200 OK reply is truly for the INVITE (or whatever other transaction is initiated by your SIP packet), it *must* have the *same* Call-ID per the RFC, otherwise it's not a valid reply. The Call-ID is what's called a GUID (Globally Unique IDentifier). It is up to every SIP user agent to generate one, and the only requirement is that it be as unique as practical in time and SIP space. Many network elements like to tack on IP addresses in the GUID as a means of differentiating it further, though personally I think that's a bad idea. Would you mind pasting a capture of the transaction in question, from the vantage point of the outside interface of your Asterisk host? You can change the representations of the external IP to something else if you don't want to post it to a public list. Thanks, -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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 Wireshark export of two packets pasted below. I simply did a find/relace and put %EXTERNIP% in place of my actual public, PATed, IP address. That is only modification I did to these pcaps. No. Time Source Destination Protocol Info 1 0.00 192.168.114.64 209.62.1.2 SIP Request: OPTIONS sip:sip.us1.voip.ms Frame 1 (544 bytes on wire, 544 bytes captured) Arrival Time: Sep 4, 2009 13:36:02.490711000 [Time delta from previous captured frame: 0.0 seconds] [Time delta from previous displayed frame: 0.0 seconds] [Time since reference or first frame: 0.0 seconds] Frame Number: 1 Frame Length: 544 bytes Capture Length: 544 bytes [Frame is marked: False] [Protocols in frame: eth:ip:udp:sip] [Coloring Rule Name: UDP] [Coloring Rule String: udp] Ethernet II, Src: Dell_95:35:26 (00:22:19:95:35:26), Dst: Cisco_7d:53:80 (00:0e:38:7d:53:80) Destination: Cisco_7d:53:80 (00:0e:38:7d:53:80) Address: Cisco_7d:53:80 (00:0e:38:7d:53:80) ...0 = IG bit: Individual address (unicast) ..0. = LG bit: Globally unique address (factory default) Source: Dell_95:35:26 (00:22:19:95:35:26) Address: Dell_95:35:26 (00:22:19:95:35:26) ...0 = IG bit: Individual address (unicast) ..0. = LG bit: Globally unique address (factory default) Type: IP (0x0800) Internet Protocol, Src: 192.168.114.64 (192.168.114.64), Dst: 209.62.1.2 (209.62.1.2) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 00.. = Differentiated Services Codepoint: Default (0x00) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Total Length: 530 Identification: 0x6abe (27326) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x08f4 [correct] [Good: True] [Bad : False] Source: 192.168.114.64 (192.168.114.64) Destination: 209.62.1.2 (209.62.1.2) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Source port: sip (5060) Destination port: sip (5060) Length: 510 Checksum: 0x0739 [validation disabled] [Good Checksum: False] [Bad Checksum: False] Session Initiation Protocol Request-Line: OPTIONS sip:sip.us1.voip.ms SIP/2.0 Method: OPTIONS Request-URI: sip:sip.us1.voip.ms Request-URI Host Part: sip.us1.voip.ms [Resent Packet: False] Message Header Via: SIP/2.0/UDP %EXTERNIP%:5060;branch=z9hG4bK69fa843c
[asterisk-users] Problem with NAT settings?: SIP reply CALL-ID from ITSP has internal address in host part
We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b1d72f1328c...@%externip% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b1d72f1328c...@%internip%. I can not figure out where the ITSP is even getting my %INTERNIP% from, I don't see if in the packet anywhere. I have externip, localnet, and nat=yes all setup in my sip.conf. Any ideas of where to look for the source of this problem? -aws ___ -- 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
Re: [asterisk-users] G729 license count...
On Fri, 18 Apr 2008 08:37:32 -0800, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: If you care to use ping pong balls and the atlantic ocean as your medium, you should be able to interface with the g729 codec if you still needed to :D I've heard that RFC1149-compliant devices work well with g729 as well :) -- Godwin Stewart - Horwich IT services ___ -- 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] Busy (congestion) signal and cell phones
On Wed, 16 Apr 2008 08:40:42 -0500, Mark Gimelfarb [EMAIL PROTECTED] wrote: why do cell phones and Gizmo both detect busy tones and terminate the call? Is that a standard behavior? It *is* standard procedure for a cellphone to terminate a call immediately it discovers that the called number is busy. It will then, optionally, initiate its auto-redial function etc. Why don't landlines do that? Because back in the old days there were no intelligent electronics to tell the user that the call failed. A special busy tone had to be generated to inform the user that they should hang the receiver up manually. Some traditions die hard. -- Godwin Stewart - Horwich IT services ___ -- 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] OT - How to check HPET is on and working before installing Asterisk ?
On Fri, 11 Apr 2008 08:40:20 +0200, Olivier [EMAIL PROTECTED] wrote: Before installating Asterisk, zaptel and so on (and independently of those), I would like to check HPET is on and working. $ zgrep HPET /proc/config.gz CONFIG_HPET_TIMER=y CONFIG_HPET=y CONFIG_HPET_RTC_IRQ=y CONFIG_HPET_MMAP=y Or, if your config is not exposed under /proc, then this: $ grep HPET /usr/src/linux/.config CONFIG_HPET_TIMER=y CONFIG_HPET=y CONFIG_HPET_RTC_IRQ=y CONFIG_HPET_MMAP=y As a last resort, if the kernel's config is available under /proc and you don't have the kernel source installed: $ grep hpet /proc/timer_list Clock Event Device: hpet set_next_event: hpet_legacy_next_event set_mode: hpet_legacy_set_mode HPET showing up as not working means a kernel rebuild. -- Godwin Stewart - Horwich IT services ___ -- 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] OT - How to check HPET is on and working before installing Asterisk ?
On Fri, 11 Apr 2008 14:32:36 +0200, Olivier [EMAIL PROTECTED] wrote: So my question remains : how can I be certain HPET is included and enabled without messing with zaptel and subsequent operations ? HPET is part of the Linux kernel. Messing with zaptel and subsequent operations is not going to get it working. If none of the tests I described reveal it then it is not included in your kernel and you need to build a new one which includes it. -- Godwin Stewart - Horwich IT services ___ -- 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] Where is the Digium DS3 card?
On Sun, 6 Apr 2008 17:22:58 -0400, Jay R. Ashworth [EMAIL PROTECTED] wrote: Yes, I've seen that, and most of its arguments are specious, at best. They amount to I am too stupid to find a mail user agent with List Reply, and too lazy to switch to it. Are there any MUAs (other than Microsoft's pitiful offerings) that do not observe RFC2369 headers? -- Godwin Stewart - Horwich IT services ___ -- 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] Where is the Digium DS3 card?
On Mon, 7 Apr 2008 11:35:43 -0400, Jay R. Ashworth [EMAIL PROTECTED] wrote: Question is: does Mailman *set* it? Yes. -- Godwin Stewart - Horwich IT services ___ -- 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] AsteriskNOW and IE
On Thu, 3 Apr 2008 09:39:55 + (UTC), [EMAIL PROTECTED] (Tony Mountifield) wrote: nothing was shown in the main pane. So there is definitely something wrong with IE compatibility. s/ compatibility// There. I fixed your post :) -- Godwin Stewart - Horwich IT services ___ -- 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] Question about PCI Slots for DIGIUMs Boards
On Thu, 27 Mar 2008 06:48:58 -0400, Al Baker [EMAIL PROTECTED] wrote: I an considering using *your High Density T*1 cards on a number of servers we are considering purchasing. The vendor lists that his system has: PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X: 64-bit/100MHz* Could you please clarify *WHICH* of the above listed *PCI slots* are suitable for use with your *High Density T1 cards*. None of the above listed are PCI slots. PCI != PCI Express -- Godwin Stewart - Horwich IT services ___ -- 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] Is Asterisk ready for Prime-Time?
On Wed, 19 Mar 2008 16:38:23 -0500, Bill Andersen [EMAIL PROTECTED] wrote: Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? /me raises hand. This said, if I did acquire sufficient knowledge of the system to be able to sell Asterisk-based solutions, I would probably do just that. -- Godwin Stewart - Horwich IT services ___ -- 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] Hardphone SIP phone costs
On Wed, 19 Mar 2008 11:32:44 -0400, Anciso, Roy [EMAIL PROTECTED] wrote: When I starting thinking about it, can anyone else see a time when desk phones are replaced by smart phones? Why would a company pay for work cell phone and desk phone when one device could potentially do it all? Definitely. I have a Nokia N95, which does precisely what you say. When I'm home (I work from home) it's hooked up to my Asterisk setup over the WLAN. When I'm out and about it's just a conventional cellphone. I know there are issues that need to be considered like safety (911) for one. But can anyone else see where I'm coming from on this. My VoIP provider doesn't do emergency calls either. Who cares? If the need arises there are 3 cellphones and the land line here as well as the N95 on which I can place an emergency call. -- Godwin Stewart - Horwich IT services ___ -- 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] Telemarketer Torture....
On Sun, 16 Mar 2008 08:50:50 -0500, Lyle Giese [EMAIL PROTECTED] wrote: I just forward them to one of those two extensions. If callerid worked more reliably I would automate it. But I get a lot of caller id failures on my incoming POTS lines, esp when calling in from my cell phone. The way I worked around this problem was to give a passcode to people I want to hear from even if they conceal CLI. If an inbound call comes in without CLI (or with CLI but the number is in my blocklist for that matter), I forward it to a recorded message saying Caller ID screening is in operation. Please press 1 if you are an authorized caller. When the user complies, they're prompted for the passcode. If it's correct, then the call is forwarded to my extension. Those I do want to hear from are not just blown off, they have a chance to get through to me regardless of the screening. Teleslime doesn't, and they've paid for the call anyway. -- Godwin Stewart - Horwich IT services ___ -- 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] Dialing patterns and GSM format numbers
On Fri, 14 Mar 2008 07:29:33 +, Adrian Merwood [EMAIL PROTECTED] wrote: Secondly (in the future) I would like to strip off certain country codes and replace them with a local dialing prefix. Can anyone help me figure ths out? This might get you started: http://howto-pages.org/asterisk -- Godwin Stewart - Horwich IT services ___ -- 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] Mail Server
On Fri, 14 Mar 2008 13:06:27 +0200, love U.all [EMAIL PROTECTED] wrote: ur mail erver isn authorized to redirect mails say for example to hotmail coz msn deal with it as spam MSN and hotmail are not a reference in anything related to Internet e-mail. Unless, that is, you're considering how *not* to form a mail body and how *not* to deal with junk mail. -- Godwin Stewart - Horwich IT services ___ -- 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] NIN Ghosts music (free download) safe for MOH?
On Mon, 10 Mar 2008 09:17:16 +0100, Dave Cotton [EMAIL PROTECTED] wrote: Ok now I am curious, if a radio is playing in a store, a restaurant or at the beach, wouldn't that be considered a public performance? From a conversation with a hairdresser who fell foul of this the answer is in France you do have to pay. Confirmed. I lived there from 1983 until a few months ago and I know for a fact that bars have to have special TV licenses in order to show, for example, soccer matches and other sporting events, and a radio license in order to broadcast the radio to clients, many of whom are too p*ssed to realize what they're listening to or watching anyway :) -- Godwin Stewart - Horwich IT services ___ -- 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] Dead Air on PF firewall
On Mon, 10 Mar 2008 07:00:17 +0800, NOC ph [EMAIL PROTECTED] wrote: I have an asterisk box on my DMZ, and I'm using a PF for my firewall, I can make a call but some reasons I have a dead air. Judging by the fact that you're portforwarding port 5060, I'm guessing that you're using SIP with the outside. This also means that you need to allow the RTP stream though your NAT FW. Port 5060 only carries the signalling, the audio is carried by the RTP stream, which is why you're getting no audio. Google will probably let you know which UDP ports your appliances are using for the RTP stream. General help that you'll be able to refine WRT the specifics of your setup is available here: http://www.google.com/search?q=asterisk+%22no+audio%22 -- Godwin Stewart - Horwich IT services ___ -- 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] Call forwarding-in india
On Mon, 10 Mar 2008 16:22:45 +0530, sandeep [EMAIL PROTECTED] wrote: Can any body tell how to enable call forward facility in INDAI for an asterisk IPPBX. Why would it be different in India from anywhere else? -- Godwin Stewart - Horwich IT services ___ -- 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] Silencing VoiceMail() app in * 1.4.10
On Fri, 7 Mar 2008 16:08:31 +0200, Mindaugas Kezys [EMAIL PROTECTED] wrote: Then you can change channel language in front of VoiceMail() app and in appropriate place put auth-thankyou file which is recorded/made by you. Much as I dislike this kludge because of the potential for b0rkage when Asterisk is updated, for now I've backed up the original auth-thankyou.gsm and symlinked silence/1.gsm to auth-thankyou.gsm. -- Godwin Stewart - Horwich IT services ___ -- 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] Silencing VoiceMail() app in * 1.4.10
On Sun, 9 Mar 2008 16:22:35 +0200, Atis Lezdins [EMAIL PROTECTED] wrote:
I think that giving 's' argument should silence all prompts including
auth-thankyou. You should report a bug on http://bugs.digium.com ,
fixing this should be trivial.
It isn't that trivial.
I've looked at the source and the silent flag is not passed all the way
down the chain to the function that actually does the recording.
In apps/app_voicemail.c, the option is parsed by vm_exec() and passed on to
leave_voicemail().
leave_voicemail(), however, doesn't pass it down to play_record_review().
So by the time the call stack goes through ast_play_and_record_full() and
__ast_play_and_record() in main/app.c, where we see the foillowing code,
the status of the silent option is long lost:
if (outmsg == 2) {
ast_stream_and_wait(chan, auth-thankyou, chan-language,);
}
I will, however, work on a patch to pass the silent option down the chain
to this function, but it's going to mean a major overhaul.
--
Godwin Stewart - Horwich IT services
___
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Re: [asterisk-users] Silencing VoiceMail() app in * 1.4.10
On Sun, 9 Mar 2008 21:49:34 +0200, Atis Lezdins [EMAIL PROTECTED] wrote: Maybe you should ask for best way for this in asterisk-dev. Good point. I'll probably do that tomorrow. I checked wat you're saying and it seems to me that more logical would be to play auth-thankyou in application, not __ast_play_and_record(), but it may break some concept. The thing is there are multiple applications that use these functions and that are liable to want to say thank you afterwards. With that in mind, it does make more sense to have the message sent out from the common denominator rather than from each application that uses that common denominator. However, I also agree with your statement that the 's' option should silence *everything* with the exception, perhaps, of the beep before recording starts. -- Godwin Stewart - Horwich IT services ___ -- 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] Silencing VoiceMail() app in * 1.4.10
Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten = 2,1,Playback(/media/asterisk/answerphone-en) exten = 2,n,VoiceMail(2000,s) exten = 2,n,Playback(/media/asterisk/thankyou-en) exten = 2,n,Hangup() The 's' option to VoiceMail() silences the prompt, leaves the beep just before going into 'record' mode, but also plays back auth-thankyou after the user hits the # key. How can I suppress playback of auth-thankyou at the end or get VoiceMail() to play back a different file? Thanks in advance, -- Godwin Stewart - Horwich IT services ___ -- 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] Silencing VoiceMail() app in * 1.4.10
On Fri, 7 Mar 2008 12:10:37 +0200, Mindaugas Kezys [EMAIL PROTECTED] wrote: Just find this file in /var/lib/asterisk/sounds and change it to anything you like. But that will break other applications that use the auth-thankyou sound, Authenticate() for a start (which I use elsewhere in order to remote check the voicemailbox). -- Godwin Stewart - Horwich IT services ___ -- 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] Newbie: confusion with the new FXO/FXS card
Or better yet, download the PDF of the Asterisk: The Future of Telephony (aka the starfish book): http://asteriskdocs.org/ Glenn Cobb wrote: Go here www.voip-info.org and read alot. Almost everything you need to know (or a link to it) can be found through there. Seriously, its your best starting point regards, Glenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vytenis Sabaliauskas Sent: Wednesday, January 09, 2008 7:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie: confusion with the new FXO/FXS card Hello everyone, I'm trying to set up a Asterisk server. I have two cards - one is an BeroNet BN2S0 with two ISDN lines (4 channels): http://www.adcomtec.com/webstore/beronet_bn2s0.php?cat=90 and a Rhino R8FXX with one FXO module and two FXS: http://www.voipsupply.com/product_info.php?products_id=2940 I would like to set up an Asterisk server with 8 phones, which will share the phone numbers via ISDN. Sorry if i'm not writing this very clearly, since my telecomunication skills are appaling (I'm used to linux though). As documentation states, Rhino R8FXX can work as FXO or as FXS depending on the modules installed. How about my situation? At first, I would like to set up a simpliest thing - to make phone do anything (play WAV, echo, etc.) Does anyone show me the road how to do it? A link to some manual or such... I tried googling, but have found no results (or didn't knew, that the info was good for me). This is my first Asterisk server and I'm very interested in bringing it up. Thanks. Ow, and sory for laminess :) -- V. ___ -- 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] X-Lite problems on basic asterisk setup
I'm trying to setup my first Asterisk setup on a CentOS 5 installation on VMWare Workstation 6. Got two Linksys SPA941s working fine. But X-Lite softphones can't answer phone calls, and when one of them calls on of the Linksys phones they connect but neither party can hear hear the other. I noticed that the Linksys phones are connected via Native bridging while the X-Lite ones are connected via Packet2Packet bridging. Also, on the X-Lite phones there is a about a 30 second lag between when the X-Lite client hits dial/call and when the called party starts ringing. ::Asterisk setup:: Asterisk 1.4.4 Zaptel 1.4.3 (only ztdummy compiled) Asterisk Addons 1.4.1 CentOS 5 VMWare Workstation 6 ::sip.conf:: [Linksys01] type=friend secret=ledzep context=default host=dynamic mailbox=6445 [X-Lite01] type=friend secret=rammerjammer context=default host=dynamic dtmfmode=rfc2833 mailbox=2070 canreinvite=yes nat=no [Linksys02] type=friend secret=bigben context=default host=dynamic mailbox=6368 qualify=yes ::extenstions.conf:: [default] include = demo exten = 6445,1,Dial(SIP/Linksys01,20) exten = 6445,n,Voicemail(u6445) exten = 2070,1,Dial(SIP/X-Lite01,20) exten = 2070,n,Voicemail(u2070) exten = 2070,n,HangUp() exten = 6368,1,Answer exten = 6368,n,Ringing exten = 6368,n,Dial(SIP/Linksys02,20) exten = 6368,n,Voicemail(u6368) exten = 6368,n,HangUp() --- Andrew Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X-Lite problems on basic asterisk setup
Packet sniffer found the problem. RTP was firewalled on the Asterisk box. Fixed it using the Asterisk firewall rules page on the wiki http://www.voip-info.org/wiki-Asterisk+firewall+rules. The 30 second lag on the dialing has something to do with using the domain name instead of the IP address of the asterisk server in the SIP config on X-Lite. The call goes immediately when I set the domain to the IP address of the asterisk box. Thanks for your help. Rob Schall wrote: This typically happens when the phone is natting or there is a firewall between the phone and the asterisk server. The connection is made via sip (5060), but the voice is over ports 1-2 (RTP). Most likely, the sip connection is succeeding, since you are connecting, but the actual voice is failing to transfer over RTP. if this is the case, I would aim to use IAX since it was made for this type of use. If the phone is on the same network as the asterisk server, and you are still having issues, use a packet sniffer and watch the traffic on both ends. You should be able to receive every packet that is sent. Most likely in this case though, you will only see those 5060 packets making it. Rob Andrew Stewart wrote: I'm trying to setup my first Asterisk setup on a CentOS 5 installation on VMWare Workstation 6. Got two Linksys SPA941s working fine. But X-Lite softphones can't answer phone calls, and when one of them calls on of the Linksys phones they connect but neither party can hear hear the other. I noticed that the Linksys phones are connected via Native bridging while the X-Lite ones are connected via Packet2Packet bridging. Also, on the X-Lite phones there is a about a 30 second lag between when the X-Lite client hits dial/call and when the called party starts ringing. ::Asterisk setup:: Asterisk 1.4.4 Zaptel 1.4.3 (only ztdummy compiled) Asterisk Addons 1.4.1 CentOS 5 VMWare Workstation 6 ::sip.conf:: [Linksys01] type=friend secret=ledzep context=default host=dynamic mailbox=6445 [X-Lite01] type=friend secret=rammerjammer context=default host=dynamic dtmfmode=rfc2833 mailbox=2070 canreinvite=yes nat=no [Linksys02] type=friend secret=bigben context=default host=dynamic mailbox=6368 qualify=yes ::extenstions.conf:: [default] include = demo exten = 6445,1,Dial(SIP/Linksys01,20) exten = 6445,n,Voicemail(u6445) exten = 2070,1,Dial(SIP/X-Lite01,20) exten = 2070,n,Voicemail(u2070) exten = 2070,n,HangUp() exten = 6368,1,Answer exten = 6368,n,Ringing exten = 6368,n,Dial(SIP/Linksys02,20) exten = 6368,n,Voicemail(u6368) exten = 6368,n,HangUp() --- Andrew Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Stewart [EMAIL PROTECTED] (205) 585-2980 - cell ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provisioning Linksys PAP2T ATA's
You can reload via http using a command like: wget\ --output-document=/dev/null\ --quiet\ http://ip-address-of-pap/upgrade?http://ip-address-of-web- server:80/asterisk/spa000F66A83C90.cfg I tried it with my xml file and it complains about the file being corrupt. I'm guessing you need Sipura's configuration compiler. I managed to talk their support people out of the compiler for the spa3k several years ago. Maybe you can pratice your SE skills. I believe that you should use the 'resync' keyword instead of 'upgrade'; the latter is intended to specify the URL of a new firmware image. I'm guessing that in some cases it looks at the file contents to decide whether it's configuration data or firmware, so it works anyway. See http://www.sipura.com/Documents/faq/Section_2.html#11 --Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 45
I will be out of the office until Monday, May 14. Please contact OWD at 800-337-3839 and ask for Client Services if you need immediate assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 46
I will be out of the office until Monday, May 14. Please contact OWD at 800-337-3839 and ask for Client Services if you need immediate assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 39
I will be out of the office until Monday, May 14. Please contact OWD at 800-337-3839 and ask for Client Services if you need immediate assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 42
I will be out of the office until Monday, May 14. Please contact OWD at 800-337-3839 and ask for Client Services if you need immediate assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] pap2 - dtmf works when 'sip debug' is enabled
I am having an odd problem with a linksys pap2 ata and asterisk... Asterisk won't detect digits from it until I issue a 'sip debug'. As soon as I turn on sip debugging, everything works perfectly (classic heisenbug)! Instead of SIP debug, try capturing the traffic with tcpdump etc. on the Asterisk server. If even that is too invasive, connect the PAP2 and a PC to the network via the same dumb hub (or managed switch); run wireshark on the PC to capture. If you catch the PAP2 misbehaving, make sure you have the latest firmware for it. If no luck, try setting DTMF Tx to AVT. If still no luck (and your network speed and jitter permits), perhaps alaw codec with inband tones will work. --Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Linksys PAP2 and Caller ID
Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to show the Caller number on the phone. There's a Caller ID Method: option on Regional settings, but I tested all options, and my CLIP phone never shows the Caller number... It should work fine. First, verify that you have for Line 1 (if phone connected to port 1): CID Serv: yes CIDCW Serv: yes and for User 1: CID Setting: yes CIDCW Setting: yes Of course, you must not answer until several seconds after the first ring completes. If you are using distinctive ring (or Default Ring is not 1), there are many subtleties that may be causing your trouble; try without it. If no luck, use a butt-set or similar to check whether CID modem tones are present after the first ring. If yes: Test the phone on a POTS line or another service to be sure CID is working ok. If so, try playing with ringing voltage, frequency, or waveform. If no: Use SIP Debug (or networking tools) to look at the SIP received by the PAP2; confirm that Asterisk is sending valid CID info. --Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 won't answer or dial. Help?
I've spent several days now trying to get my TDM400 card to work. I'm running TrixBox 1.1 (at least to start). I've tried an old PII 233mhz with 256 MB, and a modern Dell Dimension 8400 (P4 3.0 ghz, 1 GM RAM). On both machines I have a series of installation headaches, some of which seem to be TrixBox's fault, since they are fairly consistent between machines. I can get Asterix FreePBX working as long as I don't use the TDM400. It seems to work great with the soft SIP phone. I can dial into it, get voicemail, record messages, etc. But I need it to work with the TDM400 card to be useful. I've followed the Nerd Vittles and SureTeq guides approximately, and think I should have the ability to receive calls and dial out over the TDM's FXO port. When I dial in, and watch from FreePBX's panel, I see that the Zap 1 trunk goes red. The SIP soft phone I have the call directed to then gets the call. I have the SIP soft phone on auto-answer, so it picks up. Or, rather, it thinks it picks up, since the calling phone continues to ring and eventually goes to my POTS voicemail. A similar thing happens when I dial out. I call my cell phone, Zap 1 trunk goes red again, and the soft SIP phone says call established but the call never gets made on the phone line. (I can pick up with a conventional phone and get a dialtone.) It seems to me that * has recognized the TDM correctly, and that I've picked the right port on my TDM. But, something is clearly wrong, and I can't find mention of this kind of thing in the FAQs. - Here I call the inbound trunk, which Asterisk is supposed to pick up. [EMAIL PROTECTED] etc]# asterisk -r == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.9.1 svn rev 34876, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.9.1 svn rev 34876 currently running on asterisk1 (pid = 3170) Verbosity was 1 and is now 4 Core debug is at least 1 -- Starting simple switch on 'Zap/1-1' -- Executing NoOp(Zap/1-1, Entering from-zaptel with DID == ) in new stack -- Executing Set(Zap/1-1, DID=s) in new stack -- Executing NoOp(Zap/1-1, DID is now s) in new stack -- Executing GotoIf(Zap/1-1, 1?zapok:notzap) in new stack -- Goto (from-zaptel,s,7) -- Executing NoOp(Zap/1-1, Is a Zaptel Channel) in new stack -- Executing Set(Zap/1-1, CHAN=1-1) in new stack -- Executing Set(Zap/1-1, CHAN=1) in new stack -- Executing Macro(Zap/1-1, from-zaptel-1|s|1) in new stack -- Executing NoOp(Zap/1-1, Returned from Macro from-zaptel-1) in new stack -- Executing Goto(Zap/1-1, ext-did|s|1) in new stack -- Goto (ext-did,s,1) -- Executing Set(Zap/1-1, FROM_DID=s) in new stack -- Executing Set(Zap/1-1, FAX_RX=disabled) in new stack -- Executing Goto(Zap/1-1, ext-local|200|1) in new stack -- Goto (ext-local,200,1) -- Executing Macro(Zap/1-1, exten-vm|200|200) in new stack -- Executing Macro(Zap/1-1, user-callerid) in new stack -- Executing GotoIf(Zap/1-1, 0?report) in new stack -- Executing GotoIf(Zap/1-1, 0?start) in new stack -- Executing Set(Zap/1-1, REALCALLERIDNUM=) in new stack -- Executing NoOp(Zap/1-1, REALCALLERIDNUM is ) in new stack -- Executing Set(Zap/1-1, AMPUSER=) in new stack -- Executing Set(Zap/1-1, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(Zap/1-1, 1?report) in new stack -- Goto (macro-user-callerid,s,9) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing Set(Zap/1-1, FROMCONTEXT=exten-vm) in new stack -- Executing Set(Zap/1-1, VMBOX=200) in new stack -- Executing Set(Zap/1-1, EXTTOCALL=200) in new stack -- Executing Set(Zap/1-1, CFUEXT=) in new stack -- Executing Set(Zap/1-1, RT=15) in new stack -- Executing Macro(Zap/1-1, record-enable|200|IN) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060704-220403|1152075839.2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060704-220403|1152075839.2: Inbound recording enabled. recordingcheck|20060704-220403|1152075839.2: CALLFILENAME=20060704-220403-1152075839.2 -- AGI Script recordingcheck completed, returning 0 -- Executing Monitor(Zap/1-1, wav49|20060704-220403-1152075839.2| mb) in new stack -- Executing GotoIf(Zap/1-1, 0?dolocaldial|1) in new stack -- Executing Macro(Zap/1-1, dial|15|tr|200) in new
[Asterisk-Users] Re: Linksys PAP2T-NA - call goes through but phone doesn't ring
I'm trying out a Linksys PAP2T-NA. Calling out works great, no problems there. Calling in, though, the phone doesn't ring. Caller ID shows up, I can pick up the phone, and the call is connected, but no ring. I've tried it on two analog phones, same behavior. Suggestions? I don't know if the T version is different. On a PAP2, in Advanced View, Regional tab, you can tweak Ring Frequency (default 25, try 20 and 30), Ring Voltage (default 70, try 90) and Ring Waveform (default Sinusoid, try Trapezoid). Also, assuming you're not using distinctive ring, make sure Ring 1 Cadence is 60(2/4) or something reasonable. If still no luck, you may have a bad SLIC; try the other port. If all else fails, use a scope to see what ringing voltage, if any, is present. It's hard to believe that this could be a * problem. --Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as MGCP User Agent
Hi David and all, I have a voip provider that uses mgcp and I would like to connect that provider to my asterisk. Anyone succeed in doing this? I have a similar interest, for Free Télécom (France) DSL, which includes an MGCP based VoIP service. I have been too lazy to tackle this myself, but would be willing to contribute some code. Another possibility is a standalone gateway program that acts as SIP server and MGCP UA. --Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] uplink call quality issues
Hi, Are you sure that this is an Asterisk problem? Configure an IP phone, ATA, or softphone to connect directly with the provider, and check the quality. If it's bad, use tools such as http://www.testyourvoip.com/ and http://www.pingplotter.com/ to troubleshoot. If standalone phone works ok, compare with *. Same codec? Same packetization? If not, adjust * to match. If they're the same, please provide info on your Internet connection upload speed, codec, type of client phones, etc. --Stewart We are using [EMAIL PROTECTED] 1.5 and SIP trunks to communicate to the PSTN network. We are having some problems with the call quality. Although we can hear the other person's voice quite clear when making or receiving a call, we get complaints from the people on the other end saying that our voices sound very unclear, low and that the voice drops, therefore people on the other end can not understand what we are saying. But as I said in our end their voices sound clear. I have checked network wise and found no latency problems within our small LAN, with our VoIP provider and reaching their SIP server's IP address, also the CPU load in the asterisk server has been graphed and does not exceed the normal CPU load levels Any assistance will be very much appreciated PolAus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FXS or VOIP
Hi Jim, My decision had more to do with the infrastructure of the existing wiring more than anything else. I really *wanted* to go with voip but I couldn't justify the extra cost since our office is wired for analog. I ended up going with the TE410P Quad span T1 card, 2 PRIs and an adit-600 channel bank for the FXS ports. I really had to do very little to tune the FXS ports other than setting tx and rx gain on the channel bank. We have 5 other branch offices that we are connected to via WAN and we have * servers at each of those locations, doing voip between those and also the larger install that I describe above. So just because you have FXS ports does not mean that you cannot do voip. There's always services like nufone for long distance that you can connect * to. For your smaller setup just evaluate what's there already in terms of network infrastructure then decide what fits best for both your budget and your growth. Best Regards, Jason Stewart On 11/01/06 15:06 -0600, Jim Freeze wrote: Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS cards. Is there any reason why I would need to consider using analog phones and FXS cards? Seems to me the cheapest way is with voip phones and voice quality should be good since the phones are on a private network that only has voice traffic. Thanks -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] read .what else to do ?
Please note that recent IOS has SIP NAT traversal turned on by default. I believe that it only supports internal UA / external server. Since you also want the opposite, you should probably turn it off: no ip nat service sip tcp port 5060 no ip nat service sip udp port 5060 Some IOS versions will even crash on SIP behind NAT. See http://lists.digium.com/pipermail/asterisk-users/2004-January/033718.html Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP. You can edit rtp.conf to use e.g 1-10007 (would allow 4 calls) and then only 8 ip nat statements would be needed for RTP. You don't say what's failing. make calls outside our LAN sounds like you are trying to call using a VoIP provider that Asterisk registers with. But your remote SIP phones is something different; which of the above are failing? Are the registrations successful? Is it just the RTP that's not working (in which case the called phone will still ring)? If not, what error or timeout is reported? If * verbose and/or debug logs don't show precisely what is going wrong, use Ethereal (on both sides of the router if necessary) to see what is happening. --Stewart Hi all , I have tried configuring Asterisk at home to make calls outside our Lan WITHOUT any success (Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT ) To be precise i did the following (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2 Forward UDP Port 1 to 2 to 192.168.1.2 (2) I set externip = x.x.x.x (to our public WAN) localnet =192.168.1.0 /255.255.255.0 (3) I also set nat=yes qualify=yes (4)Please,I know alot of you out there have implemented AAH to work outside your network ( Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT ).Please advise me how to make it work !!! (5) I am using xten lite soft phone on my pc . (6) I use cisco 1700 series router ,and i have natting configured on this router .Maybe I am using a wrong command .Please,tell me the commands to forward the ports Port 5060-5082,1 to 2 to 192.168.1.2 on a cisco router . Please reply and advice !!! Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip man in the middle
Hi Mike, This is wanted because using to ATA back to back creates a number of problems with echo. Also a delay for CID and problems with DTMF decoding. Keep everything digital is the way to go. Agreed. But before getting started with Asterisk, I posted a similar idea to the group; it was met with a quite cool reception, on and off-list. See http://lists.digium.com/pipermail/asterisk-users/2004-October/068932.html . I ended up avoiding Vonage and using multiple other providers. That said, I believe that many users of non-BYOD ITSPs would benefit from a proxy such as you describe. Unfortunately, I'm not aware of anyone that has implemented it yet. If you undertake such a project, IMO you should do it in Asterisk, or as a separate process that can run on the same machine as Asterisk, because many more people would use it and contribute to its development. --Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no have dial tone
mgcp.conf [general] port = 2427 bindaddr = 10.22.58.222 [10.22.58.199] context=iad101e host=dynamic callerid = 169 169 nat=no canreinvite=yes line = aaln/0 extensions_additional.conf exten = 169,1,Dial(MGCP/aaln/0 at 192.168.0.22) now the problem is when i dial from a sip ext to mcgp gateway 169, the phone connected to the mgcp gateway rings and can talk, but when i call from the mgcp gateway, it no have dial tone and cannot talk,do any know this, thanks a lot. I don't know if host=dynamic can ever work with MGCP, but it can't work here, because there is no way for * to tell which MG is registering. So, try changing it to host=192.168.0.22 if that is the MG's IP. But then what is at 10.22.58.199? Are these addresses somehow related? Is there any NAT? If things are working correctly, when you restart the MG, it should send an RSIP to *, which should respond 200. Then, * should send an RQNT to the MG, which should respond 200. When you pick up the phone, the MG should send NTFY, and *, after its 200 response, should send another RQNT that causes a dial tone. Use Ethereal to check for the above, or verbose mode in *. --Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] config Polycom with both SIP provider and Asterisk
Hi, I have some SoundPoint IP 501 phones, running SIP 1.6.2. I would like to configure them so that line 1 connects directly to a SIP provider, and line 2 connects to a local Asterisk PBX. That should be simple enough, but this provider requires URIs like sip:[EMAIL PROTECTED] . However, the DNS A record for provider.com points to their Web server, and there are no SRV records, so one must supply the server IP address explicitly. There seems to be a couple of ways to do that, but I can't get either to work. In these examples, it is assumed that my number with the provider is 212333, and their server IP is 11.22.33.44 . First, I tried: phone1 reg reg.1.address=[EMAIL PROTECTED] reg.1.server.1.address=11.22.33.44/ /phone1 Unfortunately, the phone generates a URI like sip:[EMAIL PROTECTED] , and the call fails. In section 4.6.2.1 of the manual, it says For user part only registration (reg.x.address=1002), the registration will be [EMAIL PROTECTED] I would assume that when reg.1.address is *not* user part only, the supplied domain name is used, but that seems not to be the case. Next, I tried: phone1 reg reg.1.address=212333 reg.1.server.1.address=provider.com/ dialplan routing server dialplan.1.routing.server.1.address=11.22.33.44/ /routing /dialplan /phone1 This time, the correct URI is generated, but it is sent to the provider's Web server -- the routing parameter seems to be ignored. Why? Finally, I tried: phone1 reg reg.1.address=212333 reg.1.server.1.address=provider.com/ /phone1 sip voIpProt SIP outboundProxy voIpProt.SIP.outboundProxy.address=11.22.33.44/ /SIP /voIpProt /sip In that case, calls via the provider work fine, but there is then no access to Asterisk, because outbound proxy is a global setting. I suspect that the problem could be fixed by pointing the phones to a doctored DNS server that returned 11.22.33.44 for provider.com. However, we don't want the added complexity and unreliability of such a solution. Any suggestions will be gratefully appreciated. Thanks, Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sixtel
Is there anyone out there who has given this outfit money and actually received any service from them? I am about to give up on sixTel, because of poor customer service. It's a shame, because they otherwise seem quite competent. I signed up Sept. 23 at http://www.iax.cc . Outbound service worked right away, with one minor issue: calls to some busy numbers resulted in ring-no-answer. I called support and it was fixed promptly. Calls to USA and Canada are completed reliably, have good voice quality, are reasonably priced, and are accurately billed. There is also a good control interface with real-time CDRs, etc. I then signed up online for a toll-free number for testing. It works well, too, and is quite reasonable $0.019/min. plus $0.20/mo. However, my intended use for this service was to port a toll-free number that I presently have with another carrier. More than a month after submitting the resporg form, and after numerous unanswered inquiries, I received a message on Nov. 21 saying: Sorry for the delay. Your ticket request is being processed. We have been short staffed the last week or so due to an unexpected departure. It's pretty bad when one missing support person can cause so much delay. Worse, that ticket still hasn't been answered, nor have any follow-up requests. And, I can't reach them by phone anymore. The only good news is that the old carrier still has the number, so I'm not losing calls. I also have a technical issue that has gone unanswered: overseas calls don't work from my account. This is not a major problem (can use other providers), but it indicates that the lack of response is not limited to porting problems. If anyone at sixTel reads this list, and you still want my business, please answer the tickets, or reply off-list. Thanks, Stewart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: call waiting not working on PAP2 (Andy Kuo)
I have callwaiting=yes in my zapata.conf, and Call Waiting Serv: Yes in the PAP2s. However, there's sitll no callwaiting on the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Hi Andy, You also need to set CW Setting: Yes on the User 1 and User 2 screens. Or, dial from each line, whatever you have set for CW Act Code on the Regional screen. If you've already done that, or it doesn't help, please post what SIP status you get when you call the busy phone, or if you hear the call waiting tone but can't pick up the new call, what happens when you try. --Stewart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to asterisk using mgcp
im trying to make two asterisk boxes communicate on mgcp protocol only. Anybody has idea how to implement this This is presently not possible, unless you have some suitable intermediate gateway(s). MGCP is a master-slave protocol. The Call Agents control the Media Gateways. The current version of Asterisk can serve as a Call Agent, but not as a Media Gateway. This means that you can use Asterisk with MGCP phones and MGCP gateways, but not with services such as CallVantage or Free Télécom that utilize MGCP. If you find a way to make Asterisk act as a MG, please let me know, as I would like to be able to connect to MGCP providers. --Stewart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call to a particular 800 number never shows answered on Zap channel
Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. First, there is nothing unfair or illegal going on. Large toll-free users have enough clout that they can negotiate contracts, where they are not billed during the service selection phase of a call. For example, when you call American Airlines, billing doesn't start until an agent answers, or the caller selects automated flight information or a similar IVR service. Answer supervision is used to tell the carrier when to start billing. This system is quite common and used by hundreds of companies. With Asterisk, three things might go wrong: You may hear ringing instead of the initial IVR greeting. If your carrier is sending 180 Ringing instead of 183 Progress (SIP) or Alerting without inband audio (PRI), then they must fix the problem; nothing can be done at your end. You may hear the IVR answer, but can't control it, because your outbound DTMF or voice is blocked. Your carrier might be doing the blocking, in which case they obviously must fix it. However, there are also some SIP phones and ATAs that don't send outgoing audio during Progress. If you have such, adjust the configuration if possible. If not, you will need to disable reinvites. You may have two-way communication with the IVR, but the call gets disconnected before answer supervision is received. Find out if it's your carrier or Asterisk that is timing out. If the latter, just put a longer timeout in your Dial statement; 180 seconds should be enough. --Stewart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call to a particular 800 number never showsanswered on Zap channel
Is there a way I can tell if it is asterisk or the carrier that is timing out from the CLI? Sorry, I don't have PRI and don't know the details. However, I'm sure that if you set a high enough verbose or debug level, you'll see the ISDN messages between * and the carrier's switch. I don't know which terminology will be used, but you should see * send an IAM (perhaps called Initial Address Message or Setup) and the switch reply with ACM (perhaps Address Complete Message or Call Proceeding). Then, about 60 seconds later, you'll see REL (Release). Who sends it? If it's your carrier, ask them why it comes so soon. If it's *, perhaps your SIP phone is the culprit. If it's not obvious from its config, set up a local extension that doesn't time out to voice mail, call it from your SIP phone and see if it will ring for more than a minute. If neither your carrier nor your phone is timing out, then I guess it must be * but I don't know where that might be. Perhaps some * guru can help. Also, is there a way to force the phone to start the call counter or force the answer on the asterisk-side. I would guess that if you called Answer() before Dial(), then the call counter would start. However, it would also start on busy signals, rejected calls, etc. Sorry, I don't know if there is a way to have it start only when call progress is received. --Stewart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Regcontext/regexten broken??
Recently I've noticed two bits of odd behavior with respect to regcontext/regexten in CVS HEAD 1.2 Beta1, and I was wondering if anyone could shed some light on this. I've set up a regcontext in sip.conf. I've set up two users with regexten entries, one in sip.conf and one in a mysql realtime table. The first bit of oddness is that regexten seems to worksomewhat as described for users whose entries are in sip.conf, but for the user whose entries are in a realtime database, it doesn't seem to workat all. Specifically for the sip.conf user, the cli reports adding the extension upon registration, and 'show dialplan' indeed shows the added entry. For a user configured through a realtime database, the cli reports adding the extension upon registration, but 'show dialplan' shows no added extension (and indeed attempts to dial the allegedly registered extension fail). The second bit of oddness is that in the sip.conf.sample it states Patterns may be used in regexten however, while registering a sip user with regexten=_45X does yield an entry (according to 'show dialplan' for the regcontext) of '_45X' = 1. Noop(test)', attempts to dial anything that should match that pattern (451, 452, etc) in that context result in reports ofno such extension...it appears almost as if pattern matching is not being performed on extensions added by SIP. So...question is, what's broken here? Is is Asterisk? My understanding? Or my installation of Asterisk? All three...?? ;-) If anyone can shed some light, I'd greatly appreciate it. Stewart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which hardware configuration? How would this work?
Hello Everyone, Please accept my appologies - I've been reading through the handbook and the online documentation / mailing list archives and can't quite get my own answer to these inquiries... The biggest mystery is how the existing handsets are connected to a new machine running Asterisk. Background: - The phone system we have is horribly out of date and may pack-it-in any day now. - Existing PBX system (AltiReach running on NT4) but we plan on replacing this server entirely and ditching the old PCI cards but keeping the hand sets (approximately 30 Nortel hand sets). - We have 12 regular phone lines coming into this system - We have satelite offices that could be VOIP after the system is implemented. What is the best hardware configuration for this? Should we get a T1? Which cards/hardware should we use? We are currently unclear on how the hand sets connect to the system but moderately clear on how the phone lines would connect to the box. Some information sources or direct examples of how to switch from a 30 handset office to an Asterisk system would be awesome. Once we replace our current setup we will delve into the extended features/options available. VOIP is probably the most important one after we switch systems entirely. If there is anything else I can provide to help you help me I will reply as soon as possible. -- Landon StewartSuperb Internet Corporation ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does the error stale nonce' mean?
Hi Paul, I'm receiving the following error over and over, adnauseam: Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce received from 'CNAME-CID sip:[EMAIL PROTECTED]' Does anyone know what stale nonce is? Thanks! This is normally not an error. Digest authentication in SIP is very similar to its use in HTTP. See http://www.ietf.org/rfc/rfc2617.txt . Details for SIP are at http://www.ietf.org/rfc/rfc3261.txt . When your client sends an INVITE or a REGISTER, * will challenge with a pseudo-random nonce (in the 401 or 407 response), and the client will reissue the request with a corresponding digest; the request is then accepted if the digest is correct. If the client needs to reregister or call the same number again, it is permitted to supply the same digest in the new request, usually avoiding the need to send two requests. However, if * decides that the nonce is too old, it will send a new challenge, to make replay attacks more difficult. * includes stale=true in the authenticate request, to tell the client that the password was ok and it can recompute the digest without asking the user to enter new credentials. Does this happen on REGISTER, on INVITE, or both? For all clients, all of the same type, or just one device? How often? Does the client reissue the request, and does it then succeed? --Stewart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP service from Free Téléc om
I'd like to use the VoIP service from Free with Asterisk, but am having a couple of problems. Here are some details: ADSL from Free Télécom comes bundled with VoIP and TV services. Most users access the VoIP via the supplied Freebox, which is an integrated DSL modem, router, ATA, and media player. It is of course possible to connect the Freebox to Asterisk via an X100P or other FXO interface. However, to improve quality, reliability, control, etc., I'd like to have Asterisk directly access the underlying MGCP service. Since this will take quite a bit of work (chan_mgcp presently acts only as Call Agent and cannot function as an endpoint), I first tried to configure an old Cisco ATA-186 to use the Free service. Although international and domestic long distance calls (both outgoing and incoming) work fine, there are problems with local calls. When calling some locations in Paris, the ATA user hears a severe echo (though there is no echo if Freebox is used). The 186, like most ATAs, has echo cancellation only for the analog line. That is working as expected; the remote party does not hear an echo. I would think that the far side echo would be canceled by the remote media gateway, but that does not seem to be the case. I don't believe that the caller has any control over this (the CA sends out requests and the endpoint obeys them), so it appears that the Freebox must be doing echo cancellation for both ends. Can someone confirm this? If it's true, is it possible for Asterisk to cancel echo from the remote end? On calls to nearby locations, such as my own POTS line or Free's voicemail service, there is no outgoing audio from the ATA. It appears to be a routing problem, because I can't ping these media gateways, typically 172.16.254.x, but can ping those where the audio is ok, typically 172.25.x.x. Packets do arrive *from* 172.16.254.x, and incoming audio is ok. However, the ATM protocol is RFC 1483 routed, VC mux, so there is no way to specify a gateway other than using the proper PVC, which I assume is 8/35 for all the private addresses used for VoIP, and 8/36 for Internet IPs. I'd like to see what the Freebox is doing differently, but don't know how, because this traffic does not appear on its Ethernet port. Is there a reasonably inexpensive tool that can monitor the packets on a DSL line? Or some other way to find out what is happening? Thanks in advance, Stewart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 working peer to peer
Hi Luis, Can anyone can tell me if I can connect 2 Cisco ATA-186 in a peer to peer layout (without an Asterisk server registerisng the devices) through Internet? If running MGCP or SCCP, no. If running H.323 or SIP, and both ATAs are on static public IPs, no problem. Just specify the address of each unit as the gateway or proxy for the other. Disable registration. If NAT and/or dynamic IP is involved, it depends on what firmware version you are running, whether the NATs are aware of the protocol being used, and whether you have administrative control of them. But, why are you trying to do this? If you just register the two units with Free World Dialup or similar, it should work ok with NAT and dynamic IP, and the config will be provided for you. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Disable Console Audio
On 22/07/05 02:49 +0900, Kuniyoshi Murata wrote: Hi, Now, I think I want to disable Asterisk's access to console audio device based on the logic above. How can I do that? Make sure the following is in your modules.conf file: noload = chan_alsa.so noload = chan_oss.so ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Busy Extensions.
On 21/07/05 15:22 -0400, Tim King wrote: I seem to have almost everything working now. The only problem is all of my extensions seem to be busy. I can call out, but not in. Can someone point me to the settings in the extensions file that could cause this. Hi Tim, Nice to see a fellow Grand Rapidian on the list :) It looks like you're using AMP, which makes the troubleshooting process hard since we cannot ask you for your extensions.conf. Check that the extension that you're dialling is set up correctly. If you can dial out then this is probably the problem. What kind of hardware are you using for FXO? Jason Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX over HTTP
HTTP uses TCP. Too much overhead. Add SSL on to that and you have a huge amount of overhead. The end result would be poor and choppy sound quality. Jason On 21/07/05 21:58 +0200, Rob Scott wrote: For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: So you all think VoIP sypply is warm and fuzzy
On 18/07/05 17:06 -0700, Michael D Schelin wrote: I was waiting for everyone to reply so here is mine.. Check out the Mediatrix web site. There are no downloads or lists of resellers who might have this provisioning software that is normally included with purchase. You may be right that it is a refurb but every indication points that it is not. I have contacted both companies and I'm waiting for replys. I'm on the west coast and it took over 7 days to get here. I am a little pissed when all other ATA's are configurable from their built in web server. And Yes, I'm self serving as well as mostly everybody I've ran into in this business. This unit was purchased for testing. Because of the timezone problem, When I get the product from UPS it's too late to call Canada or FL. when all I need is a simple download to correct the problem. Is it too much to expect everything in the box when you purchase it? Or have a web site with these free included software so if this happens we don't wast our valuable time. By the way I did get an email from VOip Supply asking me to wait until morning so they could find the software. This is at 2:30 PST. This complaint was to hear from others about VoIP Supply and their business practices. I wanted to get feedback ether way, or maybe a contact name so I can get this paper weight working and tested. Has anyone used the 2102? Please let me know. Obviously you have a misunderstanding. Why not assume that there is a misunderstanding, with voipsupply then work from there instead of dumping your anger out on all of us? I don't doubt that there is a CD or there was once a CD that shipped with the 2102, but - According to the Medaitrix Web Site... --- Copy and paste from mediatrix web site --- With the Mediatrix 2102, service providers get the product characteristics allowing them to successfully deploy residential IP telephony applications. The Mediatrix 2102 provides a web interface, giving users a convenient access to the unit for initial set-up. The Mediatrix 2102 can auto-provision by fetching its encrypted configuration file from a TFTP or HTTP server making installation transparent to end-users. To further facilitate deployments, factory loaded configurations are possible. Automatic firmware and configuration file downloads ensure that the 2102 is always up-to-date. --- end --- You are supposed to use a web interface for initial set up. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] So you all think VoIP sypply is warm and fuzzy
I am a little pissed when all other ATA's are configurable from their built in web server. The 2102 does have a built in Web server. See manuals at support.bctgroup.ru/mediatrix/2102/ If you have a refurbished unit, perhaps the web server was disabled, or the password was changed. Try reset to factory settings. It is possible to disable the factory reset, and conceivable that the previous owner did that. However, if he did, the SNMP community string was probably also changed, and the CD that you are complaining about wouldn't do you any good. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] So you all think VoIP sypply is warm and fuzzy
The 2102 does have a built in Web server. If you have a refurbished unit, perhaps the web server was disabled, or the password was changed. Try reset to factory settings. It is possible to disable the factory reset, and conceivable that the previous owner did that. However, if he did, the SNMP community string was probably also changed, and the CD that you are complaining about wouldn't do you any good. What happens when you try to access the unit? Can you at least ping it? --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SS7
On 07/06/05 11:30 -0400, Matt wrote: Hi, Has anyone used the SS7 link from Digium? If so, how did it work for you? Any issues? Anything to be aware of? Do I just need a T1 card like the PRI card I have now from Digium? Hi Matt, There are some links to user reports on the wiki: http://www.voip-info.org/wiki-Asterisk+SS7 It also looks like your Digium PRI card will work too. If you're in doubt call Digium, I'm sure they would answer your questions. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A newbie question - SIP to Trunk
Hello, Firstly sorry for covering old ground - I'm new to this. . . . I've read that you have to be careful when configuring SIP phone extensions so that an incoming call can't be connected to the trunk. Anyone have some info on how this can happen and how to stop it? Next, Can anyone tell me (in outline) how to set up a wifi SIP phone so that when I'm in the office I call in/out over Asterisk and the trunk and when I go home I can still be called from the office and still use the office Asterisk for trunk calls. Of course the office Asterisk is behind a NAT/firewall. Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Voip Technology : RTP over TCP
I am interested in implementing RTP over TCP Why? If you want to permit operation through a firewall that blocks UDP, there are packages that provide VPN tunnels over TCP or even HTTP. You could then run any VoIP system over that VPN. As you said, delay performance would sometimes be awful. Skype will automatically fall back to TCP if a UDP connection attempt fails. Most of the commercial instant messaging packages that support voice or video can work over TCP. If your purpose is to improve performance on networks with high packet loss rates, IMHO you would get better results from a UDP-based system that permits forward error correction, by transmitting each voice frame in two or more packets. If you can't afford the increased bandwidth, a system of retransmission such as used by popular streaming protocols would still be better than TCP. One more point is What is feasibility of implementing RTP over TCP in case of NAT (Network Address Translation) is there ? Any of the above systems can work through NAT. If both endpoints are behind NATs, and you can't set up port forwarding on either, then of course you must connect via an intermediate server. Skype and the IM services do that automatically. If your desire for TCP is not related to firewalls or packet loss, I'd be interested in hearing about your application. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 186 MGCP Firmware
Hi Ken, Can't seem to find it anywhere, and my cisco login works, but says there's no longer any downloads available for the ATA186.. anyone know where I could find the MGCP version of the firmware via download? Log in. From the main page, click the dropdown list for Downloads and select Voice Software. That takes you to http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml Under Voice Applications Software, click on ATA 186/188 Analog Telephone Adaptor That took me to http://www.cisco.com/cgi-bin/tablebuild.pl/ata186 The latest seems to be ata_03_01_01_mgcp_040629_1.zip ATA Version 3.1.1 software for MGCP, 02-JUL-2004 When I clicked that link, the license agreement came up. I did not proceed, but it seems likely to work. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 186 MGCP Firmware
I've been there.. the page comes up with There are currently no files for this type. Well, you either have a technical problem or an administrative one. Eliminate the possibility of corrupted cookies or browser cache by going to another workstation, accessing http://www.cisco.com/cgi-bin/tablebuild.pl/ata186 and entering your credentials. If you still see no files listed, it appears that Cisco has (perhaps inadvertently) downgraded your account. Open a case with them. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toll Free dialing problems
I've tried using iaxtel and BroadVoice to route toll free calls and the call appears to connect ok (see log snippet below) but it just rings and rings and eventually it times out and I get The person you are calling is unavailable Hi Shadow, This is a common problem, not limited to VoIP. Large toll-free users, such as IBM in your example, have enough clout with their carrier, that they don't pay for minutes during the IVR portion of the call. This is accomplished by not sending answer supervision until the call is sent to a human. If you have SS7, there is enough information to make this work properly. If you have a dumb POTS line, there is also no problem, because the CO switch takes care of it for you. But with an interface of moderate intelligence, such as T1, or sometimes with PRI, SIP, or H.323, there can be trouble. First, verify that this is your problem. Try calling (888) 746- via Broadvoice. You should hear the call the talk line advertisement. Or, call a toll-free number that is answered by a human; it should work ok. If you have trouble with *all* toll-free numbers, see if setting pedantic=yes in sip.conf helps (using CVS HEAD). If not, post a more detailed log. However, if your problem is as described above, use Ethereal to capture and play some audio from BroadVoice during the 183 Progress. If you hear ringing, the problem is at BroadVoice and you'll have to get them to fix it, or find another provider. But if you hear IBM's IVR, then either Asterisk is not passing the audio properly to your client (IMO unlikely, use Ethereal to check), or your client is not processing the Progress correctly (test with a different SIP client or a non-VoIP extension). Once you can hear the IVR, you may have trouble getting outbound DTMF to work during Progress. Your phone or ATA may have an option send RTP during Progress or something similar. Good luck, Stewart Log snippet below: -- Executing Dial(SIP/116-3e81, SIP/18887467426 at sip.broadvoice.com|45) in new stack -- Called 18887467426 at sip.broadvoice.com -- SIP/sip.broadvoice.com-ace8 is making progress passing it to SIP/116-3e81 -- Nobody picked up in 45000 ms -- Executing Congestion(SIP/116-3e81, ) in new stack == Spawn extension (inside, 18887467426, 2) exited non-zero on 'SIP/116-3e81' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem parsing unusual SIP/SDP
The next step would to be turn pedantic=yes back on, then generate a failing call with 'sip debug', 'set verbose 255' and 'set debug 255' in place. Capture all the output (there will be a lot) and then post a bug in Mantis describing the situation and attaching the output file. Kevin, thanks again for the help. I now understand why it's not working, but don't know enough to suggest a fix, or even to say what routine has the bug. The problem relates to the additional checking done by find_call when pedantic=yes. In response to the original INVITE, the provider sends a challenge with a tag: SIP/2.0 401 UnAuthorized [other headers] f:Test User sip:[my phone [EMAIL PROTECTED];tag=as5822c02a t:sip:[dest [EMAIL PROTECTED];tag=1628255942721615 WWW-Authenticate: Digest ... [other headers] Asterisk saves the tag in the theirtag member of the sip_pvt structure and issues a new INVITE with suitable credentials. The provider initiates the call and returns progress: SIP/2.0 183 Session Progress [other headers] f:Test User sip:[my phone [EMAIL PROTECTED];tag=as5822c02a t:sip:[dest [EMAIL PROTECTED];tag=e5559e9a-1dd1-11b2-b48e-b03162323164+e5559e9a Well, provider is now sending a different tag, so Asterisk does not find a match, assumes that this response is for a call it does not know about, and discards it. Although this is ugly SIP, one can understand why it would happen, and IMHO it is legal. RFC 3261 says: When the originating UAC receives the 401 (Unauthorized), it SHOULD, if it is able, re-originate the request with the proper credentials. I believe that re-originate means that we are starting a new dialog and the old tag should be discarded. However, I don't know where or when this should be done. In fact, I don't understand why the tag checking happens on outgoing calls at all. A comment in chan_sip.c says: /* In principle Call-ID's uniquely identify a call, however some vendors (i.e. Pingtel) send multiple calls with the same Call-ID and different tags in order to simplify billing. The RFC does state that we have to compare tags in addition to the call-id, but this generate substantially more overhead which is totally unnecessary for the vast majority of sane SIP implementations, and thus Asterisk does not enable this behavior by default. Short version: You'll need this option to support conferencing on the pingtel */ That makes sense, but since Asterisk always generates a unique Call-ID for each call, I would think that tag checking on outgoing calls would be unnecessary. However, the routine carefully chooses the From or To field according to the call direction, so there is probably a good reason to check all calls. Indeed, the change that I would request might break operation with some other provider or device. Is it worth posting such a vague bug report? Unfortunately, I know absolutely nothing about the internals of Asterisk. Thanks, Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Problem parsing unusual SIP/SDP
Is it worth posting such a vague bug report? Unfortunately, I know absolutely nothing about the internals of Asterisk. Yes, please do, but make sure you include a full 'sip debug/set verbose 255/set debug 255' as an attachment in the bug. Also include the relevant portions of your sip.conf file (with secrets removed, of course). I submitted a bug report, and was amazed that within two hours, Mark had found the bug, fixed it, and posted updated source code. The code works great. There are now no problems on incoming or outgoing calls. I never get such good support for commercial software, even on high-end packages that charge an arm and a leg for maintenance. Many thanks to Mark, Kevin, and the Asterisk team. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Emailed voicemail
Richard, Yep, got that config'd in there: 1001 = 1001,Andy Stewart,[EMAIL PROTECTED] 1002 = 1002,Lorri Barnett,[EMAIL PROTECTED] 1003 = 1003,Andy Stewart - Home,[EMAIL PROTECTED] 1004 = 1004,Andy Stewart - HardPhone,[EMAIL PROTECTED] 1005 = 1005,Lorri Barnett - HardPhone,[EMAIL PROTECTED] Or it this maybe the problem? Your example is ext = ext,emailMine above (and the example in voicemail.conf) is ext = ext,name,email ?? Thanx A From: Richard J. Sears [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Emailed voicemail To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hi Andy, did you configure voicemail.conf with the users e-mail address...? 1234 = 1234,[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 8, Issue 210
Richard, I feel a little stupid now. Our spam filter (GWAVA) was blocking the emails because I had WAV files in the block list. One of those things that doesn't occur to you until you've had a little bit of sleep. Thanx for the help! A * Date: Fri, 25 Mar 2005 06:03:42 -0800 From: Richard J. Sears [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Emailed voicemail To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Yes Andy - that was my mistake. I have my system hacked up to do some other things. It should be: 1234 = 1234,Bob Jones,[EMAIL PROTECTED] do your mail logs have any errors at all in them in regards to mail bouncing or anything like that..? Do you have your servermail settings configured in voicemail.conf and did you (maybe) compile asterisk to use asterisk_vm mysql db instead of the voicemail.conf..? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Emailed voicemail
Have Asterisk us at running fine, but have run into a small snag. It's not emailing the voicemails to the users. I have attach=yes set, sendmail is configured and works from from the commandline (sent an email to myself). Unless I'm wrong, or missing something, asterisk is configured by default to send an email when a users receives a voicemail, correct? Thanx A ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem parsing unusual SIP/SDP
I'm running [EMAIL PROTECTED] (Asterisk 1.0). Is it possible that this bug
has already been fixed in a later version (I can't find anything that
seems relevant at bugs.digium.com)?
This issue (multiple c= lines) has already been fixed in CVS HEAD (if
'pedantic' SIP parsing is enabled), but the fix was not backported to
the stable branch.
Kevin, many thanks for the advice. The good news is that I got it to
work! However, there is quite a bit of bad news:
First, there seems to be something about [EMAIL PROTECTED] that is
incompatible with CVS. After I did the checkout, there were errors
at install time, resulting from an out-of-date sounds.txt file.
Deleting the file and rerunning CVS resulted in the same wrong version.
I manually retrieved the latest sounds.txt, and the install ran ok,
but there were other old files, including the chan_sip.c that I needed.
So, I deleted the zaptel, libpri and asterisk directories, and ran CVS
from scratch. Asterisk now built ok, and ran with no obvious problems.
However, when I set pedantic=yes, I can't call out at all via the
provider, though I can still call from one SIP extension to another.
It appears that the 183 Progress is not being seen by Asterisk,
because the SIP/provider- is making progress passing it ...
message does not appear, and the provider keeps retransmitting the 183.
Unfortunately, nothing in the log looks like an error report, and I
don't know how to debug this further.
So, I left out the pedantic=yes, and in chan_sip.c where it says
/* Check for Media-description-level-address for audio */
I patched:
if (pedanticsipchecking) {
to:
if (1) {
and it now works fine with this provider.
You are welcome to enter a bug in the Mantis bugtracker to request that
the same fix be put into the stable branch, but there may be some
opposition since that would potentially change existing behavior there.
I doubt that this fix would break any SDP from another device, because
RFC 2327 clearly states that a media-level c= should override a
session-level c=, and e.g. the ATA 186 follows that rule.
However, the code calls ast_gethostbyname to get the IP, with comment:
/* XXX This could block for a long time, and block the main thread! XXX */
so maybe it's not a good idea to have it on by default (in typical cases,
though, the IP would be numeric and there would be no blocking).
And, the fix would do me no good, unless the problem with pedantic=yes
also gets found and fixed somehow.
How should I proceed? IMO, this provider offers an excellent combination
of price, reliability, quality, and support, and I believe that many in
Asterisk community would want to use them. AFAICT, their SIP/SDP does
not actually violate any RFCs.
[I have not identified them, because they are not a BYOD company,
although their TOS does not prohibit alternative interface devices.
I want to get their permission first, but would like to approach them
with Your system works great with Asterisk ...]
Thanks,
Stewart
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[Asterisk-Users] Problem parsing unusual SIP/SDP
Hi, I'm testing Asterisk with a new provider. On calls to US toll-free numbers, there is no audio (calls to normal numbers are ok). In response to a valid INVITE from Asterisk, something like this is received: SIP/2.0 183 Session Progress v:SIP/2.0/UDP [my public IP]:5060;branch=z9hG4bK62d91cea CSeq:103 INVITE i:[EMAIL PROTECTED] f:Test User sip:[my phone [EMAIL PROTECTED];tag=as341d210b t:sip:[EMAIL PROTECTED];tag=b6e96dae-1dd1-11b2-a01e-b03162323164+b6e9 6dae m:sip:[EMAIL PROTECTED]:5075 c:application/sdp l:170 v=0 o=- 3459442714 3459442714 IN IP4 192.168.201.25 s=SIP Call c=IN IP4 192.168.201.11 t=0 0 m=audio 52322 RTP/AVP 0 c=IN IP4 [provider public IP] a=rtpmap:0 PCMU/8000 The 200 OK arrives with similar SDP. Note that there are two connection addresses in the SDP, one private (the provider's -- I'm not using NAT) and one public. The problem is that Asterisk attempts to send media to the private address; of course, that doesn't work. If I use the provider-supplied ATA, or a Cisco ATA, it works fine. The SDP is similar, but the ATAs know to send media to the correct public IP. At first, I thought that the incoming SDP was improper, but RFC 2327 says: A session announcement must contain one c= field in each media description (see below) or a c= field at the session-level. It may contain a session-level c= field and one additional c= field per media description, in which case the per-media values override the session-level settings for the relevant media. So, it appears that Asterisk is not interpreting the SDP correctly. I'm running [EMAIL PROTECTED] (Asterisk 1.0). Is it possible that this bug has already been fixed in a later version (I can't find anything that seems relevant at bugs.digium.com)? If so, is there an easy way to upgrade [EMAIL PROTECTED] from the CVS? If not, could someone please suggest where to start looking at the code? Thanks, Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] seeking GSM 850/1900 gateway
Hi, I'm looking for a reliable, reasonably-priced, single-channel interface between * and US GSM. The VOIP GSM Gateways listed at http://www.voip-info.org/wiki-VOIP+GSM+Gateways (VoiceBlue, QUTEX) are multichannel systems, very expensive ($2500 or more). Next step down, there are various Fixed Cellular Terminal (FCT) or Fixed Wireless Terminal (FWT) devices. These typically have an FXS interface. http://www.voip-info.org/wiki-Asterisk+Connecting+to+the+Cellular+Network Unfortunately, the mainstream devices (Nokia, Ericsson, Siemens) seem to omit the US 850 MHz band. The few that have this band, e.g. Telular, are quite expensive ( $500) and I can't seem to find any good reviews. At the bottom are docking stations for cellular handsets, e.g. CellSocket. They are cheap enough (~ $100 + an old phone), and there is lots of commentary about them, but alas, it's mostly negative. Anyone have good luck connecting to US GSM? Thanks, Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Best DB
On Tue, 2005-03-15 at 13:00 -0500, Giudice, Salvatore wrote: MySQL: Speed, Power and Precision _ Speed, yes. Anyone can write an SQL layer over a flat file and make it fast. If you want real speed (faster than MySQL with the same level of reliability choose SQLite. Power - I agree here too. There are lots of great tools for MySQL due to it's ubiquity. Precision - No Way! see- http://sql-info.de/mysql/gotchas.html MySQL is free. It can be installed in less than 59 minutes from source for light use by a first time user AND there is no need for extravagant tuning. and if you are particularly keen on undertaking elaborate tuning projects to squeeze every last drop of life from a database, you can even write your own database engine for MySQL. So a beginner user can install MySQL in less than an hour from source with no need for tuning, but if they feel the need to tune their database other than what's out of the box a newbie can write their own database engine? I'd much rather mess with a few config options that write a database engine. For the record PgSQL can be installed in the same amount of time as MySQL. For the extreme noob who knows nothing about databases and is still learning then tuning will not be a factor. For anyone else the first thing that they'll do is look at the manual for the tuning section. It's not rocket science. If you are so keen on paying for something, try buying support - MySQL AB. With PostgreSQL, you could get support from a mom and pop shop... However, either way you will save tons of money over Oracle. You could also get enterprise level support through Pervasive, a company much larger and older than MySQL AB. http://crn.com/sections/breakingnews/breakingnews.jhtml?articleId=57700307 For benchmark information comparing MySQl with several DB's on various OS's (yes Oracle and PostgreSQL are included) see the following link: http://ftp.iranscience.net/pub/databases/mysql/information/benchmarks.ht ml Hmm... More benchmarks, eh? I've see benchmarks swing both ways with MySQL being faster and others with PGSQL being faster. In my experience Postgres has handled our multi-gigabyte database much more smoothly than MySQL. Larger, complex queries seem to return much more quickly with Postgres. My mantra is pick the right tool for the job. For smaller webapps I use MySQL. For huge enterprise databases I use PostgreSQL. Regards, -- Jason Stewart | Tel: 616-532-2300 Systems Administrator/ | Fax: 616-532-3461 Programmer | Email: [EMAIL PROTECTED] Right to Life of Michigan | Web: http://www.rtl.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sorry to be a bother ISO root password
As far as I can make out the root password for the ISO download is supposed to be epping or EPPING depending upon which version you are using. I've downloaded an ISO image from the following link but neither passwords seem to work :( http://ovh.dl.sourceforge.net:80/sourceforge/asteriskathome/asteriskathome-0.6.iso any one know the password for this one? Hi Phil, http://asteriskathome.sourceforge.net/install_doc.html says that the password is password. Don't know for sure, because I haven't installed it yet. Good luck, Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk@home scary log
On 10/02/05 15:10 +0100, Jean-Louis curty wrote: so I stopped asterisk, type mail and got a strange mail saying that user [EMAIL PROTECTED] could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log Feb 9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088: to=[EMAIL PROTECTED], ctladdr=[EMAIL PROTECTED] (0/0), delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329, relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK 1107998984) the thing is i did not send any message to [EMAIL PROTECTED] nor to somebody at yahoo, anybody got the same ? what can I do ?? There's a chance that you may have been hacked, but the logs you post look more like your mailserver is an open relay. What OS/Distro are you using, what version, and do you have the latest patches applied? What services are you running? Look for strange entries with uid 0 in your passwd file. Also check for root kits with a rootkit checker (chkrootkit.org). If everything pans out security-wise then the only problem is that you MTA is configured to be an open relay. If that's the case, then you need to fix it right away before you get on umpteen million blackhole lists. Consult the docs and/or community for the MTA that you're using to fix that. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] bellster.net - GREAT advance
Sam In France, the second most important ADSL provider (named Free) Sam offers a phone line (which uses VoIP but can only be used as a FXS) Sam with unlimited free calls to landlines. I also have Free ADSL in Paris, and would very much like to get their VoIP working natively with Asterisk. Free assigns each user both a public (for Internet access) and a private (for VoIP and television) IP address, e.g. 81.57.8.9 and 10.0.8.9. For QoS, they use a separate PVC for each address (RFC 1483 routed VC mux). Using a standard (non Freebox) modem, I was able to configure an ATA-186 (with MGCP firmware loaded) to work with the Free service, both incoming and outgoing. However, I could not do this with Asterisk, because it can only act as an MGCP master, so it can't talk to Free's Call Agent, which of course is also a master. Is anyone aware of a software gateway that can act as an MGCP slave (Media Gateway) on one side, and speak IAX, SIP, or H.323 on the other? If this is not available, I would be willing to put some effort into enhancing the * MGCP stack, to also speak the slave side of the protocol. Are there other Free users that would be interested in contributing? --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729? Worth it?
The MOS (Mean Opinion Score) scale is: 5=Excellent; 4=Good; 3=Fair; 2=Poor; 1=Bad. Some values, taken from Carrier Grade Voice over IP by Daniel Collins: G.711 4.3 G.729 4.0 G.729AB3.9 GSM(full rate) 3.7 The above scores assume no packet loss, minimal delay, no echo. However, IMO such scores are generally only useful for choosing among compression codecs. If you have plenty of bandwidth and minimal packet loss, you should use G.711, not only for better quality, but because it avoids issues with conferencing, DTMF relay, etc. Also, if your ITSP has upstream routes that use a different compression scheme, G.711 avoids cascaded codecs, which sound really awful, MOS 3 for sure. If you don't have enough bandwidth to handle the desired number of simultaneous calls with G.711, you obviously need to use compression; IMHO G.729 is a good choice. If you have 1% packet loss (or packets effectively lost due to excessive jitter), then G.729 may actually sound better. Lost G.711 samples are replaced with silence, sometimes with pops at the transitions. OTOH, most G.729 implementations have packet loss concealment, which continues the previous sound, gradually fading out. With 5% loss, a good G.729 system sounds like a mediocre cellular call, but G.711 sounds terrible. There are systems that use G.711 when traffic is light, but switch to compression codecs under heavy traffic to conserve bandwidth. I don't know how/if this can be done in Asterisk. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone use SunRocket with Asterisk?
Has anyone tried SunRocket with Asterisk? http://www.sunrocket.com/ The $199/yr. plan seems like an excellent value, and most reviews have been favorable. However, I don't know if it is possible to obtain the SIP credentials, so one can bypass their gizmo. Thanks, Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa 2000 phones do not ring
When I make a call to either 706 or 707 from any phone, the phone attached to the spa does not ring. However, if I pick up the appropriate phone, the connection is made and normal conversation can take place. I had the same problem with a Cisco 827-4V. It turned out that the phones were fussy about ringing frequency, given the relatively low output voltage from the SLIC on most FXS devices. The command pots ringing-freq 50Hz fixed it. On the SPA, under Ring and CWT Cadence, you can set ring waveform, frequency, and voltage. Try increasing the voltage, and/or setting the frequency up to 50 Hz or down to 20 Hz. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: VOIP Phone Suggestions
On 15/12/04 22:53 -0600, Kevin Curtis wrote: I would recommend Uniden UIP200 phones. Great sound quality with inbuilt phone book, call logs etc works great with asterisk. I recently purchased from [1]www.qualvoip.com (they also provided me sample configuration files for asterisk). Kevin One gripe about these guys - They clearly use * for their PBX product, which looks like it's not much more than * with a web based config interface. There's not one mention of * on their site! No, there's nothing wrong with that legally but they should be giving props to * instead of promoting it as their PBX software. Instead of calling the product The Asterisk Based PBX System they call it The Open System Based PBX System. Are they afraid that potential customers will discover * and try to do it on their own? Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: very OT - basic newbie networking
Is NAT enabled by default on Fedora core 1 (latest patches) ? Sorry, don't know. I believe that if you have disabled iptables by e.g. /etc/init.d/iptables stop then NAT should be off, but it still wouldn't hurt to check the source address reaching the phones. The target machines can be pinged from the * box, but not the phones. See if you can do ping -I 192.168.6.10 target IP --Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT] Adit 600 Question
Hi, I'm using an Adit 600 Channel Bank with *. I love it and it works really great for my FXS lines. One problem that I have with it (It's really not a problem yet, but it's a potential one) is that I've scoured the manaual for the Adit to see if there's a way to dump out a config file from the bank so in the event of a power and battery failure I don't have to type in the configuration commands, just load a file. Is there a way to get a config from the Adit 600 and load it back in again? Thanks, Jason Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] very OT - basic newbie networking
I have a * box with 2 nics in the following setup: Internet | 192.168.5.253 (firewall) | 192.168.5.xxx network (gw 192.168.5.253) | 192.168.5.10 (* nic 1) 192.168.6.10 (* nic 2) | 192.168.6.xxx network The netmask for both networks is 255.255.255.0 The 192.168.6.xxx networks has a 48 port switch solely for the use of cicso 7940 phones, the 192.168.5.xxx is for the pc's (winxp) / servers (2003) etc. I want to be able to access the phones (telnet/web etc) from the .5.xxx network, and I want the phones to be able to access the .5.xxx network. 1. Make sure IP forwarding is on. 2. Turn off iptables (at least for testing). 3. From a windows command prompt: route add 192.168.6.0 mask 255.255.255.0 192.168.5.10 4. Try to ping 192.168.6.10 from Windows. If it fails, recheck 1 and 2 above. If ok, try to ping a phone. If that fails, make sure phone has 192.168.6.10 as its default gw. If ok, you should now be able to access the phone's web server from the Windows box. 5. To avoid having to add a route to every Windows box, add a static route to your firewall, specifying that 192.168.6.0/24 is reached via the LAN interface using gw 192.168.5.10 . Good luck, Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: very OT - basic newbie networking
However, even though I've added the 192.168.6.10 as the gw for the 192.168.6.xx network, the phones cannot access the 192.168.5.xx network (or the internet). Well, if you can open a TCP connection from 192.168.5.xx to 192.168.6.xx, then routing in the reverse direction must be working. If you can't connect from 192.168.6.xx back to 192.168.5.xx, two things come to mind: Your * box might be acting as a NAT (aka IP masquerading) router, rather than a normal router. When you connect from a host on 192.168.5.xx to a phone, verify that the source IP seen by the phone is 192.168.5.xx . You can do this with debug features in the phone, by running Ethereal on * on the 192.168.6.10 interface, or with an external monitor. If you see 192.168.6.10 as the source address, then you are running NAT and need to disable it. The connection might be blocked by a software firewall on the destination host, e.g. Windows Firewall, on by default in XP SP2. Note that a service enabled with Local Subnet scope won't be accessible from the phones. If it's neither of the above, you'll just have to debug it. Run Ethereal on the 192.168.5.10 interface, and check for SYN packets going out and responses coming in. Accessing the Internet from the phones is another story. First, do you need it? If you are coming into * in SIP and going out to a provider or another * in IAX, * will have to proxy the call anyhow, so Internet access is not required. If both sides are SIP, and you want to get the performance benefits of reinvite, then you can try to get it working. Your firewall needs to have a static route for 192.168.6.0/24 with gw 192.168.5.10 , and it also must know to perform NAT on packets coming in from 192.168.6.xx . Some routers will do this automatically, some need a configuration setting, and with others you're out of luck. In the latter case, you could tell the router that the LAN subnet is 192.168.4.0/22, and set up * to do proxy ARP. Once you have NAT and the static route configured, you should be able to plug a PC into the 192.168.6.xx net and browse the Web. But whether you can make phone calls through this system is a complex issue. NAT traversal for SIP is often problematic, and many on this list have had to set canreinvite=no. Regards, Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA186 V2.15.ms upgrade
Hi Rodney, I dont have a cisco acount yet can some bady hel me with the ata18x-v2-16-030401a-1.zip file. You will need a PC running Windows. 1. Unzip it. 2. Read the text file ata186us.txt 3. Follow instructions in it :) This will convert your ATA from MGCP/SCCP to H.323/SIP . --Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA186 V2.15.ms upgrade
Hi Rodney, I dont have a cisco acount yet can some bady hel me with the ata18x-v2-16-030401a-1.zip file. ftp://ftp.rekom.ru/pub/ata18x/ You will need a PC running Windows. 1. Unzip it. 2. Read the text file ata186us.txt 3. Follow instructions in it :) This will convert your ATA from MGCP/SCCP to H.323/SIP . --Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP
I haven't found any recent information on this, but can Asterisk act as a MGCP UserAgent? I wish I was wrong, but I think right now MGCP in Asterisk is CallAgent only. http://www.voip-info.org/tiki-index.php?page=Asterisk%20MGCP%20channels Any other ideas for interacting with an MGCP provider? You could, of course, connect an MGCP ATA to FXO port(s) or device(s). That solution degrades quality, increases delay, may have echo problems, etc. However, it's an easy way to get started, e.g. if you have a spare ATA-186 that you can load some MGCP firmware into. I am seeking a proper solution to the same problem, as my ISP in France, Free Telecom, bundles MGCP service at very aggressive rates (including free calls to fixed phones anywhere in France) with their ADSL service. I have looked at some SIP - MGCP and H.323 - MGCP gateways, but they only talk the Call Agent side of the protocol. If you have found a solution, please let me know. If not, perhaps we could work together to write one. One possibility is enhancing MGCP support in * to allow it to act as a User Agent. Another is a stand-alone script, e.g. in perl, that would do SIP - MGCP. I'd be open to other suggestions, too. Thanks, Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice
are they really /unlimited/ in the truest sense of the word ? US$24.95, even if it's only for unlimited calls to Malaysia (where i am) seems very, very attractive. when something is this attractive, i start looking for the catch. AFAIK, no one offers truly unlimited service. Companies differ greatly in openness of the contractual details. I subscribe to unlimited (POTS domestic US) long distance from SBC. The contract clearly states that monthly usage exceeding 5000 minutes is billed at $0.04 per minute. Not cheap, but it won't break you if go a little over. At the other extreme, there are many horror stories of Vonage customers whose service was terminated, without warning, for excessive usage. Broadvoice appears to be somewhere in between. I am considering their service, and called them to ask about allowed usage. They would not disclose their limits, but when I mentioned that my calls typically run 2500-3000 minutes per month, mostly to the US, they said that this was well below their alarm levels. There may be a technical problem with Broadvoice for your application. I suspect that all calls proxy the media stream through their server (in the US). Perhaps a Broadvoice customer can confirm or deny this. If that's the case, the roundtrip delay on your calls to Malaysia will include *four* hops across the Pacific (~400 milliseconds). If there's any echo, it will be very disconcerting. Even if not, you'll have problems when both parties start talking at about the same time. You can use their free trial offer to see if the delay is bothersome. --Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Old Dialogic Hardware Questions
On 09/11/04 16:13 -0500, Matt Gibson wrote: Hi Everybody, I have a quick question regarding some old Dialogic hardware. We have an old Artisoft PBX (http://www.artisoft.com/PBXPhoneSystems.html). In this box are some older ISA Dialogic cards. My question is, does anyone know if the following Dialogic cards work with asterisk or in Linux at all? They are not mentioned on the digium site as supported, nor could I find anything specific to these cards on the mailing list archives. Dialogic D/80SC-4LS and Dialogic MSI/240SC-Global Thanks in advance, Matt Hi Matt, There's not many people using Dialogic cards with *. The best way to know if the Dialogic card has * drivers is to call Digium since they wrote the drivers. Be prepared to pay money for the drivers since Digium had to pay Intel to develop the drivers. I've worked with the MSI boards and I do know that you can use them with SR5.1 of the Dialogic SDK. 5.1 is the last release that Intel released for free. The MSI board is legacy and IIRC it's not supported in the newer releases. I would assume that the D/80SC-4LS is also supported. If you decide to use the Dialogic SDK be prepared to be locked into a redhat 7.2/7.3 only configuration with lots of back-assward kernel patches and STREAMS subsystem add-ons. Getting frustrated with the lack of support and rising prices on the part of Intel, I sold all my Dialogic equipment on ebay and bought a TE405P. Regards, Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP via Wireless Ethernet Bridge and Double NAT
Anyways, found an unsecured wireless network going through my new townhouse at 30% strength. Found the owner and they said I could share it for a couple of weeks. They have a Netgear, 108mbs 802.11 b/g. So I took a LinkSys WAP54g and put it in Ethernet bridge mode, it took the signal and converted it to Ethernet for me. I then plugged it into my Belkin 4 Wireless Router w/ 4 port switch. So now I'm redistributing the connection in my townhouse. I plugged a Cisco ATA-186 into the Belkin, but it's having problems registering with the Asterisk server. I figured the double NAT was messing it up. I'm getting less than 1% packet loss to the internet, so the link is strong. Cable Modem -Ethernet- Netgear Wireless Router -802.11- LinkSys WAP54G -Ethernet- Belkin Router -Ethernet- Cisco ATA186. I keep seeing sip registration failed requests on Asterisk. I checked and double checked the passwords, its fine. I believe it's that the device gets the UDP packets through to the Asterisk server fine, with the authentication information or whatever; but when the Asterisk server tries to respond via UDP, it doesn't make it through. So it fails. You can eliminate the double NAT by disabling the DHCP server on the Belkin, changing its LAN IP to not conflict with anything on your neighbor's LAN, and plugging the WAP54G into a LAN port on the Belkin. Leave the Belkin's WAN port unconnected. The Belkin should now be acting as a switch and wireless access point; it won't be doing any routing. Your computers, if set up for automatic addressing, will get them by DHCP from the Netgear. You may want to give the ATA a static address so you can forward ports to it on the Netgear. That address, of course, should be in the subnet of your neighbor's LAN, but outside of the range assigned by DHCP. Make sure that your neighbor's kids won't be hacking into your system ;) Good luck, Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * and Verisign SIP-7 service
Can you expand ... I'm hoping to use Asterisk to do SIP to MGCP mediation ... what does it not work? I don't know the particulars, because I've never used (or even looked at MGCP). All I know is that whenever the issue comes up, people here say that Asterisk does not know how to act as an MGCP Gatekeeper, only as an agent. I presume it would have to act as a gatekeeper to control an MGCP-based media gateway, because those devices are all intended to be controlled by some sort of softswitch. IMO, there is no such thing as an MGCP gatekeeper; try that phrase with Google and it will be obvious. Gatekeeper is an H.323 term. MCGP is a master-slave protocol. The master is referred to as a Call Agent, a Media Gateway Controller, or just a softswitch. This is the role that Asterisk can play. The slave is a Media Gateway, an MGCP phone, an MGCP ATA, or just an endpoint. Asterisk cannot presently act as a slave. Of course, any large system may have higher-level elements that handle authorization, accounting, complex routing, queueing, etc., but those topics are beyond the scope of MGCP. Perhaps the term gatekeeper was used in that context. So, I think that Asterisk will provide the functionality that you desire. However, I don't know if SIP-MGCP calls can presently be completed without Asterisk proxying the media stream, so you may have performance issues. Perhaps someone else can address that. --Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 3, Issue 410
i have a audio problem between sip and h323. First my installation: Debian Sarge Asterisk 1.0.1 Gnugk 2.0.8 Asterisk register a prefix to gnugk. Communication from sip to sip and h323 to h323 is working. When i now call from the siphone (three tested) the h323 phone (also three tested) the connection is coming up and everything seems to be ok (no errors, no debug info). But there is no audio in both directions. Also when i call voicemail, i hear nothing one the h323 phone. I have tested different codecs. Has anybody a hint for me, where to continue my search for the problem? Greats, Andre Peitz I assume, since you didn't mention it, that there are no NATs or firewalls in the path. I would think that H.323-voicemail would be easiest to debug. Run Ethereal on the Asterisk machine. Is * sending audio? If not: did * send a Connect, and did the Open Logical Channels happen correctly? If yes: Is it sending to the correct IP, correct port, using the right codec and correct payload size? If any routers are involved (including gnugk proxy), run Ethereal at the H.323 phone to be sure that the audio is really getting there, and that no unwanted packet mangling is happening. Can you configure an H.323 phone to call * directly (without a GK)? Also, try turning Fast Start on (or off). Likewise with H.245 tunneling. Good luck, Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
