similar to: SIP DID routing

Hello - I currently have 10 DID's coming into one Asterisk server, I seem to be having some difficulty routing based on the DID dialed and am hoping someone on the list can assist me. Here's the relevant info: Ingress SIP trunk: IP: 123.45.45.3456 DID's XXX-XXX-XX00-XX10 sip.conf: [general] useragent=Asterisk port=5060 context=default tos=lowdelay disallow=all allow=ulaw

I thought maybe one of the providers here could answer this question. When using IAX to make calls, it passes the called number. Being designed to work with a PBX this makes sense. SIP works differently though and I'm curious why providers don't have a way to pass the called number on their DID's, or choose not to do so. Providers that themselves use upstream SIP proxies are

Hi Dmitriy and others and thanks for your help so far. The option "match_auth_username=yes" seems to have had no effect. From my reading, this option will try to match the username of the incoming SIP account to a section heading. If that is how it must work then i can see a big problem. I'm trying to present the receptionist with a nice display of which line the call came in on.

I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part

Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider?

Hello, this is Jesse with Webformix. We are managing an Asterisk installation for residential VOIP service, and we are having a problem where all inbound calls to DIDs which are assigned to us by our wholesaler but not yet assigned to a downstream customer get caught in a routing loop. ** For example: we get assigned DID block 1230-1239 and only 1230-1233 are given to our customers, then our

Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj.

Hi, I just started with setting up a new asterisk system, that will operate on a sip trunk, but I wonder, how to transfer the calls to different extensions, because all calls appear as being send to the base number of the trunk. E.g. given the trunk range of 1234567800-12345678099, a call to 1234567800 is matched by the same pattern as a call to 12345678099. ; matches 12345678099, too exten

Hello all, I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts with the same service provides. We have 8 phone numbers in total. Incoming calls from the public are all correctly directed to appropriate office handsets. However, the display on the reception phone (the only one i care about) is always showing the same "SIP/Account1_0843214321" rather than the account

Hi, I'm looking for a multi-line sip/ip phone that can answer multiple incoming paths. IE a secretary sits at a front desk and can answer multiple incoming lines/DID's. Is there something like this? Thanks, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041202/5b685d69/attachment.htm

I know I can get free DID's with SIP, is anyone giving out free DID's with IAX? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Asterisk 1.4.12 I wish to extract some custom headers from a SIP REFER message but am unable to do so. However I can extract them from an INVITE. The code is: exten => _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ; exten => _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ; Examples of the INVITE (works) and REFER (doesn't) messages are below. U 147.202.001.001:5060 ->

I have a pretty strange problem. I have about 100 DID's that come down a PRI from SBC in the United States. On Friday afternoon, one of my DID's flipped out. When you call it, the SBC operator comes on and says that the line has been disconnected. I contacted them and they ran test and they are telling me the problem has to be on my end. My problem is that the CLI never shows the

I remember seeing something somewhere that described how I could get SIP header information with Asterisk. It was a command or a variable. Anyone know what it is? Thanks. Doug.

Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184 at CallFromQueue-000007f4;2 to 'Local/23 at from-internal' (thanks to SIP/myaccount184-00003729)

I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

Hi, Here is a general question, not applying to asterisk so much, but in the application of asterisk. I have purchased a few IAX DID's through VoicePulse and am interested in a service provider who has the ability to provide me with one number (reliable, as I wish to publish), and the capacity to redirect those calls to my IAX DID's (is this even possible)? Also, with IAX