similar to: CALLERID(ani2) inserting

Two years ago we added logic to parse the isup-oli parameters, that arrive as part of the FROM Sip header. We need to finish the job and allow setting of this parameter for outbound calling, both in traditional SIP channel and PJSIP, which I believe will replace all instances of the old SIP channel soon. Right now, if we try to set CALLERID(ani2)=$ {var} , there is a runtime error because this

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

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.

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"

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

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

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]:

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

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

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.

Hi I'm using asterisk 1.8 Does anyone know how to use the queue reload command. The built in help doesn't really help. queue reload {parameters|membe Reload queues, members, queue rules, or parameters Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk

Hi Is there any way to set the presence state of a peer to in-use in asterisk 1.8? The idea is to integrate DND buttons on phones to BLF. Regards -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street

Hi I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP. However, we exclusively use the asterisk realtime architecture using the mysql connector. Looking at tutorials we have to set encryption=yes and transport=tls for any peer we want encrypted traffic for. Having a look at contrib/realtime/mysql/sippeers.sql from the source code shows that the encryption column is

> > >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

Hi I'm using asterisk 1.8.23.1 but I've seen this same issue in previous versions of 1.8. I have created some work arounds but the behaviour is incorrect. This is the scenario: Call comes in and goes to appropriate dialplan In the dialplan the call is forwarded to another number using a Local channel (and using /n ) e.g. Dial(Local/<my-number>@outbound-context/n,60) The number is

>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 >>

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

Hi All, I have a question about the Queues. I'm using Asterisk 11.13.0 , and I want to configure the following setup : When there is an incoming call to the queue all agents should ring even those that are already in call, they should receive a second call. Is this doable in any Asterisk version ? Thanks in advance. -------------- next part -------------- An HTML attachment was

We use the ringall strategy for a small queue with 4 members. When a call comes in, if one of the members is busy, all the phones except the busy phone rings (as intended). While the other phones are ringing, if this busy phone becomes available again, we would like to have it start ringing. Right now it just sits idle. Is this possible? I played with ringinuse (queues.conf) and callcounter