similar to: caller id setting on channel originate

I am using AMI to Originate a call. I have been able to get the caller id number to be passed through. However, I can't get the name to be passed through. A person I'm working with has a Freeswitch that is able to pass the caller id name and number through for their call. Comparing the Asterisk SIP messages to the Freeswitch SIP messages, I have narrowed the problem down to a single

Aloha, I'm finishing up the final touches on this install, and have run into an odd problem. I can't seem to get Caller ID on the analog phone lines working. It's a Digium AEX 410 card. I have Verbose set and a line to print the CID: I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf and users.conf [analog] include=>default exten =>

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

Hi Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com> wrote: > An

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI,

Hello all, I would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: "123 <123>" Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1 at

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

Hello everyone. I am new to the list, so forgive me if this has been discussed already. My configuration is as follows: Fedora Core 3, fully patched through yum through today. Dovecot 0.9.9.13 Postfix 2.1.5 Using Maildir with Mail stored in ~/Maildir My issue is that, for some odd reason the inbox file index gets corrupt, There is no rhyme or reason as to why it happens or when, but a

I am trying to block/hide outgoing caller id on a PRI E1. It seems that it should be fairly simple, but it is defeating me. https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says: "to hide your caller id, use: Set(CALLERID(num-pres)=prohib)" That doesn't seem to do it. The calls are originated from AMI and I have tried a blank "CallerId:" line and

Hi Jöran, Would it be possible to see an example using curl of how you are passing the PAI Header through ARI create? Dan From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] With

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear caller-id dtmf tones. Pl tell me the procedure to upload recorded file if you needed. Something I want

One of our client facing this issue, we have try to solve it but we're lack of asterisk knowledge. Anybody can help us? Isn't any problem with asterisk configuration or the problem come from PRI E1 itself? [Apr 11 15:32:48] VERBOSE[9231] chan_dahdi.c: -- Requested transfer capability: 0x00 - SPEECH [Apr 11 15:32:48] DEBUG[6888] channel.c: Avoiding initial deadlock for channel

System have been working great for weeks, using an average 40 of 120 dahdi channels. But today, I suddenly see scary things like this: -- Moving call from channel 5 to channel 7 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing

So, tweaking configs, rebuilding this and that... restarting, twiddling, it works (yeah!), but fails on re-boot to work at all. Consistently, though. I believe it comes down to this: I can call out only *after* I've received a call. So, cold boot. Then: modprobe dahdi modprobe wctc4xxp modprobe wcfxo dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.3

Hi all, So, we reverted the LibPRI version and tested it, and then tried with the latest version of everything. Still no changes. The BRI line is in PTMP. If we set the configs to PTMP in the genconf_parameters and try it, we get the following: [Jan 21 17:32:20] ERROR[20341]: chan_dahdi.c:12645 dahdi_pri_error: Unable to receive TEI from network! If we set it to PTP (which it is not) we

Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel variables when I originate a call via AMI. All the documentation I have found says to do it like this: Variable: variable1=value|variable2=value|variable3=value However when I do this it runs them all together and I end up with:

Hi, I have the following happen to me after the restart of one of my servers: out of my 3 PRIs (all configured with the same technical settings), the last one isn't coming back. It's underutilized (chances it didn't get a call since my reboot), if it makes a difference . The PRI goes from provisioned to unprovisioned, and I get this regularly: [Jun 22 09:03:48]