RE: [Asterisk-Users] audio message delivery
Darren Wiebe wrote: I've been looking at a similar problem. Mine is slightly different but it involves a customer phoning in, leaving a recording, and having that recording delivered to a list of users. I hope to code at least some of this in the next few weeks. Darren Wiebe [EMAIL PROTECTED] Dean Collins wrote: Hi, I have a client who has asked me to look into the delivery of 30 second audio messages to a list of opt-in customers. Probably looking at about 5,000 messages a week over a 6 week period. I know that this would be a piece of cake to have someone develop but I thought I would ask here first if someone is already doing this and what they would charge to take this on as a hosted solution rather than having us develop it from scratch in house. Rather than coding anything, why not use take advantage of the fact that Asterisk already can do email message delivery and combine that with a good mailing list manager like ezmlm-idx (http://www.ezmlm.org/)? ezmlm will take care of all of the list management for you, and message delivery can be accomplished by creating a voicemail box that delivers to the mailing list address. If you make the list moderated then you can also approve all messages before they go out. Or do you mean audio message delivery to a list of phone numbers? --TWH ___ 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] Small office setup with Asterisk @home, IAX and analog termination
I'm setting up a small office with about 8 SIP phones. Incoming and outgoing lines will be through IAX. We would also like to use an analog line for 911. Is the TDM01B a good option for this kind of configuration? Are there gotchas I'm missing? Finally, we would like to be able to use analog fax machines in the office. Would it make more sense to purchase the TDM400 card with 1 FXO and 1 FXS port and use the FXS for the analog faxes, or to split the analog line before it goes into the Asterisk box? Or something else entirely? Thanks! --TWH ___ 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] Asterisk @home with IAX termination...
I've got an Asterisk @home setup that mostly works. I'm using Polycom SIP phones and have PSTN termination through an IAX provider. I say mostly works because, if I'm on the phone, any additional incoming call goes straight to voicemail rather than signaling call waiting on the phone. I've tried dialing *70, but that doesn't seem to fix things. I've googled the list, and checked the wiki, but I didn't see anything that seems quite like the problem I'm having. I'm sure a more informed search term would probably get me the answer right away. I'd appreciate any ideas. Thanks! =) --TWH ___ 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] Make voicemail use Maildir...
What would be involved in modifying the Asterisk voicemail system to use Maildir for storage? My thinking is that if voicemail were delivered to Maildir it would be easy to configure an IMAP server to work with Asterisk, and that would provide pretty good voicemail/email integration (where deleting a message in email deletes it from the voicemail system, etc). Thoughts? I know something like this has been discussed before but the conversation never really seems to be resolved. --TWH ___ 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: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme
Gilbert Abboud wrote: I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk through SIP. Can you please send me the Dial-peer configuration that creates a trunk between the Cisco router and Asterisk. You can try something like this: dial-peer voice 900 voip destination-pattern 9... session protocol sipv2 !(the address of the Asterisk server) session target ipv4:192.168.0.100 !(in Asterisk use dtmfmode=rfc2833) dtmf-relay rtp-nte codec g711ulaw --TWH ___ 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] Incoming calls from Cisco 1760 given wrong context...
I've installed Asterisk from the Asterisk @home distribution. Ultimately I will be using Asterisk for voicemail for about 150 users. Calls are (mostly) handled by a legacy PBX although we do have a couple of Cisco 1760 routers that connect a remote office. I've setup a SIP trunk that routes calls from Asterisk to the 1760, and that works fine. I've also configured one of the 1760s to route certain calls to Asterisk. However, the calls are placed in the from-sip-external context that Asterisk @home uses for unidentified SIP calls and are subsequently dropped. I can make the calls connect by modifying the from-sip-external context, but I would like to be able to specify that calls from the router (on a static IP) are placed in a different context. Here is part of my sip_additional.conf: [Cisco1760_mc] type=friend host=192.168.0.254 context=from-pstn disallow=all allow=ulaw allow=alaw nat=no canreinvite=yes qualify=yes However, when I use sip debug to monitor an attempted call (711515 from one of the phones connected to the Cisco), these lines appears in part of the debug: Found no matching peer or user for '192.168.0.254:53464' Looking for 711515 in from-sip-external Shouldn't it match Cisco1760_mc? I've included the full debug below. Thanks in advance for your help. I'm happy to provide any additional information. --TWH Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D From: Tim Howell sip:[EMAIL PROTECTED];tag=49582394-1813 To: sip:[EMAIL PROTECTED] Date: Tue, 15 Mar 2005 17:12:08 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 838919162-2494304729-2389872912-2937628716 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY , INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: Tim Howell sip:[EMAIL PROTECTED];party=calling;screen=no;pr ivacy=off Timestamp: 1110906728 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 267 v=0 o=CiscoSystemsSIP-GW-UserAgent 6054 4992 IN IP4 192.168.0.254 s=SIP Call c=IN IP4 192.168.0.254 t=0 0 m=audio 16946 RTP/AVP 0 100 19 c=IN IP4 192.168.0.254 a=rtpmap:0 PCMU/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:19 CN/8000 a=ptime:20 20 headers, 12 lines Using latest request as basis request Sending to 192.168.0.254 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 19 Peer audio RTP is at port 192.168.0.254:16946 Found description format PCMU Found description format X-NSE Found description format CN Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing) , combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x2 (gsm), combined - 0x0 (noth ing) Found no matching peer or user for '192.168.0.254:53464' Looking for 711515 in from-sip-external list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D From: Tim Howell sip:[EMAIL PROTECTED];tag=49582394-1813 To: sip:[EMAIL PROTECTED];tag=as2a29cbba Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.254:5060 -- Executing AbsoluteTimeout(SIP/192.168.0.254-094a8648, 15) in new sta ck -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/192.168.0.254-094a8648, ) in new stack Transmitting (no NAT): SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D From: Tim Howell sip:[EMAIL PROTECTED];tag=49582394-1813 To: sip:[EMAIL PROTECTED];tag=as2a29cbba Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.254:5060 == Spawn extension (from-sip-external, 711515, 2) exited non-zero on 'SIP/192 .168.0.254-094a8648' -- Executing AbsoluteTimeout(SIP/192.168.0.254-094a8648, 15) in new sta ck -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/192.168.0.254-094a8648, ) in new stack == Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/192.168. 0.254-094a8648' asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D From: Tim Howell sip:[EMAIL PROTECTED];tag=49582394-1813 To: sip:[EMAIL PROTECTED];tag=as2a29cbba Date: Tue, 15 Mar 2005 17:12:08 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 9 headers, 0 lines Destroying call '[EMAIL PROTECTED]' asterisk1*CLI sip no debug SIP Debugging Disabled asterisk1*CLI ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo
[Asterisk-Users] Cisco DTMF problem...
I've recently setup Asterisk. Calls are routed to Asterisk through a Cisco 1760 router. Some calls originate at Cisco 7960 phones connected to the router, some originate at other phones that are switched by a legacy PBX. My problem is that calls that begin at the 7960s do not seem to transmit DTMF properly to Asterisk. Calls that begin on the legacy PBX (and then pass through the Cisco router) do transmit DTMF. I've seen a lot of references to this in the list at various times (and even some patches), but all of the patches/fixes I've seen seem to be against a version of rtp.c that is very different than what I've got (fairly recent CVS). Are there any current fixes for this problem? --TWH ___ 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
