vicidial.org

by **fotofx** » Thu Oct 27, 2011 3:51 pm

I have my bandwidth.com carrier setup and working for outbound dialing. When calls come inbound I am getting:

Code: Select all: NOTICE[2952]: chan_sip.c:15566 handle_request_invite: Call from 'bandwidth' to extension '+13054000005' rejected because extension not found.

I have the DID setup as 13054000005 and have an extension logged in that that DID is pointed to.

Using VERSION: 2.4-334a
BUILD: 110831-2038

Installed from a clean vicibox server install

by **williamconley** » Thu Oct 27, 2011 8:34 pm

1) when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

2) what is your "context=" for this carrier? (it should be context=trunkinbound)

by **fotofx** » Thu Oct 27, 2011 8:47 pm

Code: Select all: [bandwidth] disallow=all allow=ulaw port=5060 canreinvite=yes dtmfmode=rfc2833 host=216.82.224.202 outboundproxy=216.82.224.202 qualify=300 type=friend nat=yes

So I am guessing the context is "bandwidth"

Thanks

by **williamconley** » Thu Oct 27, 2011 8:52 pm

nope. context for inbound calls is set by "context=" if that is missing it is not set and won't work.

add "context=trunkinbound".

by **fotofx** » Thu Oct 27, 2011 9:12 pm

I added the context line as directed but it is still not working. Here is the SIP debug (I changed the server IP to xxx's)

Code: Select all: [Oct 27 22:07:43] <--- SIP read from 216.82.224.202:5060 ---> INVITE sip:+13054000005@xx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 Record-Route: <sip:216.82.224.202;lr;ftag=gK0c20f72f> Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0c20f72f> Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK2a.090b7774.0 Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK2a.bce94d85.0 Via: SIP/2.0/UDP 4.55.10.97:5060;branch=z9hG4bK0cBfbc0748b51b6c329 From: "Unavailable" <sip:+19548173686@4.55.10.97:5060;isup-oli=61>;tag=gK0c20f72f To: <sip:+13054000005@67.231.4.93:5060> Call-ID: 1074574054_111663562@4.55.10.97 CSeq: 18872 INVITE Max-Forwards: 67 Contact: "Unavailable" <sip:+19548173686@4.55.10.97:5060> Content-Length: 302 Content-Disposition: session; handling=required Content-Type: application/sdp Remote-Party-ID: "Unavailable" <sip:+19548173686@4.55.10.97:5060;isup-oli=61>;privacy=off;screen=yes P-Asserted-Identity: "Unavailable" <sip:+19548173686@4.55.10.97:5060;isup-oli=61> v=0 o=Sonus_UAC 7033 28214 IN IP4 4.55.10.97 s=SIP Media Capabilities c=IN IP4 4.55.10.66 t=0 0 m=audio 14816 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 <-------------> [Oct 27 22:07:43] --- (17 headers 14 lines) --- [Oct 27 22:07:43] Sending to 216.82.224.202 : 5060 (NAT) [Oct 27 22:07:43] Using INVITE request as basis request - 1074574054_111663562@4.55.10.97 [Oct 27 22:07:43] Found peer 'bandwidth' [Oct 27 22:07:43] Found RTP audio format 0 [Oct 27 22:07:43] Found RTP audio format 8 [Oct 27 22:07:43] Found RTP audio format 18 [Oct 27 22:07:43] Found RTP audio format 101 [Oct 27 22:07:43] Found audio description format PCMU for ID 0 [Oct 27 22:07:43] Found audio description format PCMA for ID 8 [Oct 27 22:07:43] Found audio description format G729 for ID 18 [Oct 27 22:07:43] Found audio description format telephone-event for ID 101 [Oct 27 22:07:43] Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) [Oct 27 22:07:43] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Oct 27 22:07:43] Peer audio RTP is at port 4.55.10.66:14816 [Oct 27 22:07:43] Looking for +13054000005 in trunkinbound (domain xx.xxx.xxx.xxx) [Oct 27 22:07:43] <--- Reliably Transmitting (NAT) to 216.82.224.202:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK2a.090b7774.0;received=216.82.224.202 Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK2a.bce94d85.0 Via: SIP/2.0/UDP 4.55.10.97:5060;branch=z9hG4bK0cBfbc0748b51b6c329 From: "Unavailable" <sip:+19548173686@4.55.10.97:5060;isup-oli=61>;tag=gK0c20f72f To: <sip:+13054000005@67.231.4.93:5060>;tag=as084b91c6 Call-ID: 1074574054_111663562@4.55.10.97 CSeq: 18872 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Oct 27 22:07:43] NOTICE[2952]: chan_sip.c:15566 handle_request_invite: Call from 'bandwidth' to extension '+13054000005' rejected because extension not found. [Oct 27 22:07:43] Scheduling destruction of SIP dialog '1074574054_111663562@4.55.10.97' in 6400 ms (Method: INVITE) [Oct 27 22:07:43] <--- SIP read from 216.82.224.202:5060 ---> ACK sip:+13054000005@xx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK2a.090b7774.0 From: "Unavailable" <sip:+19548173686@4.55.10.97:5060;isup-oli=61>;tag=gK0c20f72f Call-ID: 1074574054_111663562@4.55.10.97 To: <sip:+13054000005@67.231.4.93:5060>;tag=as084b91c6 CSeq: 18872 ACK Max-Forwards: 70 User-Agent: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0

[/code]

by **williamconley** » Thu Oct 27, 2011 9:23 pm

Now you need to modify the extension to remove the +

Looking for +13054000005 in trunkinbound

or modify "trunkinbound" in "extensions.conf" to catch ALL extensions (or at least those starting with +) and push them to the agi script.

by **fotofx** » Thu Oct 27, 2011 9:27 pm

I thought that silly + sign was my problem. I have the DID set up without it (vici will not allow it in the did)

How do I get rid of the + sign so my DID's will work properly?

Thanks for all the help..

by **williamconley** » Thu Oct 27, 2011 10:46 pm

easiest method is in the configuration at the provider. tell 'em to stop sending the + (usually in a control panel somewhere)

otherwise, the ability to strip the first character or ignore the first character has been hashed a few times on the forum.

another method is to usse extension "s" which is ordinarily used then the real extension cannot be found (fails to "s", so you put in that extension, pointed to the same agi script and *poof* even failed extensions land there!)

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