search for: 1xxxxxxxxxx

...-- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rvvvvv): == Using SIP RTP CoS mark 5 -- Executing [xxxxxxx at phones:1] Goto("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "1xxxxxxxxxx,1") in new stack -- Goto (phones,1xxxxxxxxxx,1) -- Executing [1xxxxxxxxxx at phones:1] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "oldcidnum=0") in new stack -- Executing [1xxxxxxxxxx at phones:2] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "CALLERID(name)...

I am planning on dabbling with some VOIP providers. I was thinking of Teliax first. My thinking is that the first LD call would go to teliax and the second (etc.) calls would go out to the PSTN. I could then verify bandwidth and quality to decide when to add more trunks and to Internet connections. I have been doing some concept testing with FWD for toll free calls, but I am using 393 as a

...rouble connecting two * boxes together via SIP. It looks like * is authenticating the hostname, not the username. The Sip.conf looks fine on both sides, but I get: Jun 2 14:44:52 WARNING[2407]: chan_sip.c:6829 handle_response: Forbidden - wrong password on authentication for INVITE to '"+1XXXXXXXXXX" <sip:+1XXXXXXXXXX@206.80.70.56>;tag=as562b672b' -- SIP/+1ZZZZZZZZZZ-03e2 is circuit-busy == Everyone is busy/congested at this time -- Got SIP response 481 "Call Leg Does Not Exist" back from XX.XXX.XX.XXX The X's are the call-from number, in this case it is...

...==== [general] port=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 allow=g723 context=from-sip-external ; Send unknown SIP callers to this context callerid=Unknown register=1XXXXXXXXXX:PASSWORD@sphone.vopr.vonage.net:5061 [vonage] username=1XXXXXXXXXX type=friend secret=PASSWORD port=5061 nat=yes host=sphone.vopr.vonage.net fromuser=1XXXXXXXXXX fromdomain=sphone.vopr.vonage.net dtmfmode=rfc2833 auth=md5 [vonage_inbound] username=1XXXXXXXXXX type=friend secret=PAS...

Hi, I've been trying to use the FaxGateway application to send T.38 out over Zaptel using asterisk but I don't seem to be having any luck. I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number]) Has anyone had any luck using this thing and can enlighten me on how it's supposed to be used? Thanks.

...ething to do with a call coming in as SIP and going out as SIP. At first I thought it was a call forwarding issue because I got this message in the console: [Feb 26 12:35:19] NOTICE[1143][C-0000025d]: app_dial.c:958 do_forward: Not accepting call completion offers from call-forward recipient Local/1XXXXXXXXXX at default-00000013;1 So I put this in my dial plan: 1AAAAAAAAAA => { NoOp(${CALLERID(num)}); Ringing; Set(CHANNEL(musicclass)=none); Dial(${OUTBOUND-TRUNKR}/1XXXXXXXXXX,30); Voicemail(198,u); }; So basically as soon as someone calls incoming number AA...

...complains about rejecting a non-primary audio stream. Could this be a problem with 1.8 not liking the second media stream or is there some more configuration tweaking to be done? --- CUT ---------------------------------------------- <--- SIP read from UDP:192.168.1.13:5060 ---> INVITE sip:1XXXXXXXXXX at 192.168.1.11 SIP/2.0 From: Biscom <sip:418 at 192.168.1.13>;tag=86c9140-d281eac-13c4-55013-1f571-33180d4a-1f571 To: <sip:1XXXXXXXXXX at 192.168.1.11> Call-ID: 73cb2e8-d281eac-13c4-55013-1f571-7ef285e4-1f571 CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-1f571-7a6c24...

...try to dial out through X-Lite, I get a: Called Failed: 503 Service Unavailable Asterisk shows the following: -- Executing SetCallerID("SIP/1234-7c6e", "xxxxxxxxxx") in new stack -- Executing Dial("SIP/1234-7c6e", "IAX2/user:pass@switch-2.nufone.net/1xxxxxxxxxx") in new stack -- Called user:pass@switch-2.nufone.net/1xxxxxxxxxx Jan 24 22:05:23 WARNING[9154]: chan_iax2.c:5996 socket_read: Call rejected by 198.22.67.70: No authority found -- Hungup 'IAX2/198.22.67.70:4569/1' == No one is available to answer at this time (1:0/0/0)...

Greetings, I have a very basic equal-weight ring-all queue set up in queues.conf: [sales-queue] ;music = default strategy = ringall periodic-announce-frequency = 20 announce-holdtime = no timeout = 15 maxlen = 0 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/dude,1 member => SIP/homie,1 member => SIP/fellow,1 But for some reason, the calls to the outside SIP parties never seem to go out, if they ever did before. I've been running 1.4.x for a long tim...

...t=1 [trunk3] host=xxx.xxx.xxx.xxx port=5060 type=peer allow=ulaw dtmfmode=rfc2833 canreinvite=no reinvite=no nat=no fromuser=+xxxxxxxxxxx call-limit=1 Here's asterisk output when someone dials out: Executing [xxxxxxxxxx at from-internal:1] Macro("SIP/6001-007e2840", "trunkdial|+1xxxxxxxxxx") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/6001-007e2840", "SIP/trunk1/+1xxxxxxxxxx") in new stack [Sep 10 09:06:52] ERROR[16253]: chan_sip.c:3192 update_call_counter: Call to peer 'trunk1' rejected due to usage limit of 1 -- Couldn't...

I have a weird problem with call using my T1 card. I can make calls fine using my analog and IP phones, but when I try to initiate a call using a .call file, I get the following error -- Attempting call on DAHDI/g1/1XXXXXXXXXX for s at test:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 127 received it happens on certain numbers I dial, but if I dial that same number with an ip or analog phone that use the T1 channel, the call is going through normally. Anybody knows why...

...tml But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to add the digit "9" to what the user dials 1xxxxxxxxxx, the Polycom should actually send 91xxxxxxxxxx to Asterisk.

When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=>"SIP/1XXXXXXXXXX@sip.broadvoice.com", 'Context'=>'mycontext', 'Exten'=>'899', 'Priority'=>1, 'Callerid'=>'whatever')); It creates a screech sound when the first audio file is played. Doesn't seem to happen with another VSP I tried, but...

In the process of turning up a new pri. Zttool indicates the T1 is ready with no alarms. asterisk*CLI> pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3

...sion: DTA version 1.0 US (8x8 001111)) onto it via TFTP, so I could access the SIP configuration. Under the SIP config, I put the IP of my * system, the 5060 port, and for Domain Name, I put default (is that right?). I checked off the Send Registration Request box. Dial Plan I left at the default, 1xxxxxxxxxx|x.T (is that right?), and Transport is set to UDP. For Line 1, I have it set as follows: Phone Number=8006, CallerID Name=8006, Port=5060, AEC On=Off (no idea what this is), Username=8006, Password=1234. In the OOB Signalling page, where I set the RFC2833 options, I haven't changed anything fr...

Hi all- My TDM long distance is provided by MCI. We use account codes where MCI sends a challenge tone after receiving 1NXXNXXXXXX. Anyone have any suggestions of how to accomplish this? I can't get the soft phones to send the DTMF (the other digits go down the d-channel of our PRI). I also have not bee able to get the dial or the outgoing queue command to work. Anyone run into this?

...eceiving an incoming call on that DAHDI channel. I looked through asterisk as well as kernel logs, didn't find much. Here's some CLI outputs. Does anybody have some suggestions what might have gone wrong? ================================================ asterisk says: -- Executing [311xxxxxxxxxx at 3001-outbound:1] Dial("SIP/3001-00000a02", "DAHDI/31/1xxxxxxxxxx") in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/3001-00000a02' status is 'CHANUNAVAIL' ================================================...

I am going to have to find a fix for this problem or I'm going to have to quit using iconnect. About one call in 10 or so, iconnect's gateway gives me an error (console output appended below). So upon receiving the error, which as a 4XX error means, "Fatal," asterisk gives up and drops the call. But not iconnect!! The phone at the other end starts ringing, and rings

Other than having stripping out IPs this is what I am receiving for my voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works fine calls that come in on my PRI. BUT at least from this VOIP source the To field which is my RDNIS information for these calls, doesn't actually fill into ${CALLERID(rdnis). But as you can see I'm getting the information. My question is, Does

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further