similar to: New Feature CALLERID(ani2) read/write

Displaying 20 results from an estimated 9000 matches similar to: "New Feature CALLERID(ani2) read/write"

2015 Jan 22
1
CALLERID(ani2) inserting
I checked https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information But I cannot find a way to insert CALLERID(ani2), which I can read, but when I try to set it for a new call, I get a runtime error. This information, known as isup-oli comes embedded in the From header,like this <sip:9087311878 at 64.45.157.104:5060;isup-oli=62>;tag=sansay1724414rdb124 and it can be read
2009 Jul 20
0
Error: Invalid SIP message - rejected , no call id
On about 25% of inbound calls to a ring group, picking up any one extension as it rings results in dead air. Some details regarding my VoIP network to make the following logs more readable: 192.168.7.130 resolves to the trixbox host. 192.168.7.135 resolves to endpoint 812. 192.168.7.137 resolves to endpoint 811. 192.168.7.138 resolves to endpoint 810. 192.168.7.139 resolves to endpoint 813.
2007 May 23
0
ITSP that honors Dial Around Compensation
All, I am trying to find a SIP ITSP that honors dial around compensation. We are adding a Flex ANI code to our outgoing SIP invites by appending an isup-oli tag to our From: address, like this: INVITE sip:18889996563@carriers.icall.net SIP/2.0 Via: SIP/2.0/UDP xxx.y.34.201:5060;branch=z9hG4bK7f314484;rport From: "Dougs Payphone"
2005 Aug 24
0
ANI2 AKA Info Digits not supported?
I'm not receiving ANI2 (info digits) on my SBC PRI's. SBC said they're sending them. I called Digium support and was told it is not supported. Is anybody receiving ANI2 on a PRI? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST Newline pagesteve@sedwards.com
2013 Jan 04
0
T38MaxBitRate issue on fax passthrough
Having an issue with receiving faxes, but when I pass through the fax. Currently, I receive the fax with Digium's Fax for Asterisk, store it and the initiate an outbound call to our fax server. (XMedius Fax). This works, but we would prefer to have Asterisk simply route the call directly to the fax server and take the store and forward out of the equation. When I do that, however, the
2016 Feb 19
2
No matching endpoint found for incoming call from SIP trunk
Hello, I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway. I am able to make calls outbound through the gateway, but I am not able to make calls into the PBX from external PSTN. Specifically, an incoming call is _received_ by Asterisk, but it is not able to route the call internally owing to the following error: [Feb 18 21:08:47] NOTICE[4606]:
2016 Feb 19
2
No matching endpoint found for incoming call from SIP trunk
Thanks George, for your mighty quick response. I made the changes (re: server_uri_pattern etc.) and still, no luck--it fails for the same error. BTW, there is nothing for transport (but this is the same config from my SIP/UDP + Twilio days, which worked): *CLI> pjsip show transport twilio-siptrunk Unable to find object twilio-siptrunk. *CLI> pjsip show identifies No objects found. I did
2009 Dec 02
2
Variable Name needed
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
2009 Dec 02
0
FW: Variable Name needed
It might be worth mentioning the voip call is coming from a number we have thru bandwidth.com in case anyone uses them. James Shigley From: James A. Shigley Sent: Wednesday, December 02, 2009 3:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Variable Name needed That wasn't it either. I tried a few other likely fields from
2009 Aug 12
2
call drops after a few seconds
I have setup my asterisk box using freepbx. I can call extension and make outbound calls. the outbound calls drop between 10-30sec. we are using bandwidth.com and they have logged our call. below is your bad followed by what they say is a good call. I can't figure out where the problem is on your end. I know we are missing some stuff at the bottom but I don't know where to start.
2003 Jun 25
1
Code some * examples for me? I'll pay you! :)
Due to time constraints, I'm looking to pay someone (by paypal) for a working or/almost-working asterisk skeleton of the examples listed below... SYSTEM INFO: ---------------------------------------------------------------- I will have a single channelized T1 with all lines being available for dialing in, using E&M wink, with ANI2*ANI*DNID being sent to me (as DTMF tones I guess?). I
2003 Nov 14
0
RE: Aculab SS7/ISUP
> > >On Thu, 2003-11-13 at 16:50, Freddi Hansen wrote: > > >>>> >Freddi Hansen wrote: >>> >>> >>>>> >> with boards from Aculab, we are replacing Aculab boards with Digium >>>>> >> boards BUT we would need more >>>>> >> Digium boards IF we could use both Digium and Aculab cards in
2004 Apr 10
0
Nothing to do? Go bounty-hunting!
Being bored to death by these long weekends with nothing to do? **** Why not go bounty-hunting? **** There are some feature requests in the bug tracker with monetary bounties attached. * Windows manager * FreeBSD Zaptel drivers http://bugs.digium.com/bug_view_page.php?bug_id=0000847 * IAX incoming/outgoing limit * 2B channel transfer on PRI * MGCP media gateway support All of these have
2008 Oct 01
1
Unknown dict module: mysql
Hi I'm trying to set up the dict/expiry plugins, but the dict server always tells me it can't find the modules although they should be there and compiled in. System is: Centos 5.2 64 bit, Using rpm from http://atrpms.net/dist/el5/dovecot/ dovecot --version 1.1.3 Relevant config: dict { quotadict = mysql:/etc/dovecot-dict-quota.conf expire = mysql:/etc/dovecot-dict-expire.conf }
2003 Nov 13
1
RE: Aculab SS7/ISUP (new subject)
>Freddi Hansen wrote: >> with boards from Aculab, we are replacing Aculab boards with Digium >> boards BUT we would need more >> Digium boards IF we could use both Digium and Aculab cards in the same >> server. The reason being that >> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards >> in the servers that must support >>
2012 Jan 26
1
adding additional information to histogram
Hi, I am a beginner with R, and I think the answer to my question will seem obvious, but after searching and trying without success I've decided to post to the list. I am working with data loaded from a csv filewith these fields: order_id, item_value As an order can have multiple items, an order_id may be present multiple times in the CSV. I managed to compute the total value and the
2006 Apr 20
5
Noobie problems with helper
I have the following helper method in application_helper.rb: def format_date(date) day = to_s(date.day) month = to_s(date.month) time = to_s(date.time) date = day + "/" + month + " - " + time return date end I am trying to call this method in a view like this: <%= format_date(bounty.created_on) %> create_on is a timestamp in mysql. I am
2004 Sep 28
2
SMDI Bounty - where?
I am the one that placed the bounty. After it being there for 2 months and getting no takers (and very few if any people asking about it), we are almost finished writing it in house. I'll keep the bounty up untill we do finish our product so if anyone beats us to getting it working they'll get paid... W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com >
2002 Oct 11
0
[tcng] sysVinit script
hi below is sysvinit script for handling tcng-script .... please excuse my bad bash skils, correct me where i''m doing stupid things :") what is left... many things, if i have time i will implement them too : - start [devices] - and then correct handling of service lock files i.e. per device lock file - stats blah ... - all ideas are welcome !! - test or some such, i mean the
2013 Jun 22
2
SIP Trunking Mantra (Origination)
Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP