[Asterisk-Users] pbx_spool - outgoing qcall failure upon call progress
Does anybody have a work around for this problem. I create a call file in /var/spool/asterisk/outgoing and Asterisk picks it up and starts placing the call. However if the called channel provides any sort of progress indication (such as a SIP or IAX channel indicating ringing that causes the console to say SIP/ is ringing) the code in pbx_spool.c indicates a call failure and drops the call with reason 3. I suspect the call progress handling code is newer than the pbx_spool.c and pbx_spool.c expects any indication returned to be a failure, which is not necessarily the case anymore. Is this a bug? -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/778|20) in new stack -- Called 778 -- SIP/778-2431 is ringing -- Local/[EMAIL PROTECTED],1 is ringing Jun 6 14:39:44 NOTICE[15746]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 3 == Spawn extension (default, 571, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' ___ --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] Cisco PIX and Asterisk
It works fine for me. I have a handful of Cisco 7960s behind a PIX firewall and they register to a Asterisk server outside of the PIX with no trouble at all. I didnt do anything special to the PIX (i.e. no access list entries). The tricks I found to make it work generally apply to any setup where the clients are behind NAT. I also run the tftp server for the phones to get configs inside the firewall, and the SIPDefault.cnf file specifies the proxy address outside of the firewall. In the Cisco phone config I have these NAT settings: nat_enable: 1 ; 0-Disabled (default), 1-Enabled nat_address: ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled And the sip.conf entry for this peer is: [7000] type=friend nat=yes qualify=yes context= secret= callerid= host=dynamic canreinvite=no dtmfmode=rfc2833 timer_register_expires: 120 Setting the registry timer to 120 seconds causes the phone to send out a packet at least every 2 minutes which will open a UDP xlate on the PIX for the session. Then the trick is to use both nat=yes and qualify=yes so Asterisk chats with the phone pretty often. The interval of OPTIONS or REGISTER messages between Asterisk and phone definitely needs to be shorter than the PIXs UDP xlate timeout or the PIX will close the xlate and you wont be able to pass packets into the phone for an incoming call. Note that you can put a numeric value after qualify= instead of yes to fine-tine the interval at which it sends a OPTIONS message. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: Saturday, September 25, 2004 8:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco PIX and Asterisk I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) Internet PIX (NAT) Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the incoming to work? Has anyone managed to get this to work or am I wasting my time on this? Ta. ___ 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] Seattle IAX Termination
Packetwest Communcations provides local IAX termination service in Seattle. I use it locally for a small Asterisk setup and they provide me with DID's in the 206 NPA. They also provide outbound long-distance at rates similar to NuFone. I've had a really good experience with service quality and reliability so far. Contact them at (206) 838-4810. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Muiz Motani Sent: Friday, April 02, 2004 2:47 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Seattle IAX Termination Does anybody know of any commercial providers of IAX termination with DIDs in the Seattle, WA area? I believe the area codes are: 425, 206, 253 Failing any commercial providers, is there anybody in the seattle area running Asterisk with a PRI coming in who might be willing to sell me an IAX trunk with a DID in Seattle? -- Muiz Motani Intelligent Distribution 72-6800 Lynas Lane, Richmond, B.C. V7C 5E2 email: [EMAIL PROTECTED] phone: +1 604 448 9293 fax: +1 604 448 9296 ___ 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 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] Seattle IAX Termination
Their base rate is $35/mo per peer (single call transit at any given moment) and this provides unlimited local and inbound calling. If you are connecting a PBX and need 1 voice path at any given moment you can discuss different pricing arrangements for your needs. DID numbers are 15 cents/number/month. Long distance is something around 3 or 4 cents/min, I don't recall exactly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: Monday, April 05, 2004 12:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Seattle IAX Termination On Apr 5, 2004, at 12:18 PM, Mark Hagler wrote: Packetwest Communcations provides local IAX termination service in Seattle. I use it locally for a small Asterisk setup and they provide me with DID's in the 206 NPA. They also provide outbound long-distance at rates similar to NuFone. I've had a really good experience with service quality and reliability so far. Contact them at (206) 838-4810. Wow, a VoIP company that's even harder to contact then NuFone. Impressive :-). What kind of rates do they charge for DID numbers? Scott ___ 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 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] Auto connect to voicemail
What do you do when $CALLERIDNUM of the caller isnt the 4-digit extension? I set all of my users Caller ID entries to their 10-digit phone # so that Caller ID appears correctly when I send their call out the PRI to the public network. The side effect of this is breaking convenient access to voicemail using this method, and I havent found a way to fix it yet. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glenn Dalgliesh Sent: Monday, April 05, 2004 3:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto connect to voicemail I think this is what you are looking for Exten = 1000,1,Answer,1 Exten = 1000,2,Wait,1 Exten = 1000,3,Voicemailmain([EMAIL PROTECTED]) - Original Message - From: Mitchell S. Sharp To: [EMAIL PROTECTED] Sent: Monday, April 05, 2004 5:27 PM Subject: Re: [Asterisk-Users] Auto connect to voicemail On Mon, 2004-04-05 at 15:57, Brian Rathman wrote: I have the voicemail setup working in that I get the MWI and it emails themessage correctly. When I pressed the MWI button on my SNOM 200, it dialsinto the voicemail system and prompts me for a mailbox and password. I knowthere is a way to automatically connect directly into the mailbox via theextension.conf file, but I can not find the documentation I am looking forin reference to variables and macros for the extensions file. Can someoneplease help me with this issue?Thanks,Brian Brian, At the CLI, type 'show application VoiceMailMain'. You can use the CLI 'show applications' command to list all available apps. If you hit tab, it acts just like BASH's auto complete. Wonderful feature! Mitch Sharp Innovative Solutions
RE: [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0?
Use a soft phone as an endpoint. There are a variety of SIP and IAX softphones you can use to place a call through your Asterisk box over IP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Zheng Sent: Sunday, February 01, 2004 7:34 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] how to dial and accept a call with only x100p card on Redhat linux 9.0? Hi, all I only have x100p card, and my pc doesn't have sound card. How can I make a call and accept a call with x100p on my pc through the telephine line? If these tests are passed, I can get more money to buy a TDM 400 card and a sound card, even an IP phone. Any one can help me? I installed Redhat Linux 9.0 on pc. Thanks in advance. Best, Michael Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it!
[Asterisk-Users] Dial app does not indicate ringing to calling party
I hope somebody has seen this before...
I'm trying to use a Dial command on a inbound call to ring multiple
destinations.The calls come in to me from the provider on IAX2, and one
of the destinations I try to ring is a IAX2 to call to my cell phone.
When I add the IAX2 destination into the Dial command, the setup I am trying
to achieve works (i.e. my Zap, SIP, and cell phone all ring) but the calling
party does not hear any ringing indication... just dead air until something
answers the call (voicemail or a human).
My IAX2 provider sends me all 10 digits so I have a context set up to catch
it and divert the call to my extension (7000) for handling:
exten = 2065475023,1,SetCIDname(DID 5475023)
exten = 2065475023,2,Goto(stations|7000|1)
Ext 7000 is:
MARK=Zap/2SIP/markIAX2/[EMAIL PROTECTED]/1206xxx
exten = 7000,1,Macro(stdexten,7000,${MARK})
stdext macro:
exten = s,1,Dial(${ARG2}|30|r)
exten = s,2,VoiceMail2([EMAIL PROTECTED])
exten = s,3,Hangup
exten = s,102,Voicemail2([EMAIL PROTECTED]) ;voicemail busy path
exten = s,103,Hangup
The problem seems to key on the fact that I'm dialing another IAX2
destination during handling of the inbound IAX2 call. If I dial ext 7000
from inside (SIP or my Zap device) I hear ringing. If I remove the IAX2
destination from $MARK so that it only tries to ring local devices, it
works.. the calling party hears ringing indication as well.
Is there something I'm missing about IAX2 to IAX2 dialing like this?
Thanks!
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[Asterisk-Users] X100P modify DTMF tone length
Is there a way to get my X100P card to dial slower on the line? Mine seems to dial the digits too short/fast for the switch to catch all of the digits and roughly 25% of outbound calls fail to complete.I've monitored on the line with a test set and the audio sounds clean, and I hear the X100P send out digits really fast, but sometimes the switch seems to time out and issue a standard recording. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HOW TO GET REGISTER WITH NUFONE??
Email [EMAIL PROTECTED] to get setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alvaro Parres Sent: Sunday, October 05, 2003 2:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] HOW TO GET REGISTER WITH NUFONE?? Hi all... How can i register wit nufone i was serching at its pages... and I never find how to get register... Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IVR Questions?
Title: Message Look into AGI, and the associated AGI interface perl module. The interface is super easy to use in perl to gather digits and perform your own logic. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Sunday, October 05, 2003 5:54 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IVR Questions? I'm fairly new to Asterisk, but I've been searching the archives extensively and haven't seen much information on using IVR for more than menus. I'd like to prompt a caller to enter his ID (employee, customer, account, etc). The business use I have in mind requires a five-digit ID. Then I need to be able to capture the ID entered, validate it (probably by playing it back with say-digit) and then store the five-digit ID into a variable that I can pass to another program. Obviously, what we're aiming for is CTI. Once I have the Customer ID, I should be able to grab that customer's information from a database and pass it along with the call to the next available agent in a queue. I've noticed that its pretty easy to incorporate perl scripts into Asterisk, but I haven't seen an answer to capturing dialed digits. Thank you for your help!
RE: [Asterisk-Users] Meridian Option 11 and asterisk
You can interface the two systems a variety of ways... the quick/easy route is a direct T1 from one switch to the other. You could do it via analog trunks too, but T1 signalling makes it so much smoother overall. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Farrell Sent: Wednesday, September 24, 2003 3:03 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meridian Option 11 and asterisk Has anyone ever interfaced a merdian option 11 and asterisk. Just wondering how you went about, it's for a small setup me only need between 4/6 channels, I was thinking about using some spare ISDN channels between the two. Has anyone seen an SIP option for the meridian? European Museum Of The Year 2002 The Chester Beatty Library http://www.cbl.ie/ DISCLAIMER: The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN BRI hardware
Hi, Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm thinking of getting a BRI in my house to deliver more advanced signaling to my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux. Is there any particular BRI card that works better with Asterisk than any other? Also, can the BRI interface cards participate in conference, etc., since they aren't a Zaptel interface? Thanks, M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186 / FXO card problem
Hello, I've got a Cisco ATA 186 from Vonage plugged into my Asterisk box with a X100P card. This works great for the most part, but I'm having a disconnect supervision problem. I suspect the Cisco device doesn't provide any sort of analog disconnect supervision when it gets a SIP BYE message indicating the far-end has hung up. This causes Asterisk to leave the channel up indefinitely sometimes, if the call was in an app that doesn't time out eventually. Does anybody know of a trick to make either Asterisk deal with the Cisco's lack of disconnect supervision, or a way to make the Cisco ATA-186 provide this signaling? Thanks, Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
