similar to: Cisco PRI to Asterisk & CallerID

Does anybody have callerid name coming in on a Cisco PRI via a Cisco gateway via SIP to *? I've seen a few people ask and a few people that say it should work, but I've never seen an actual working config. I do a debug on our Cisco gateway and I can see the callerid name, however none of the features that should send it via SIP seem to work. Cisco docs say to use the following:

Does anybody know if SetCallerPres works on calls via SIP through a Cisco gateway? We are trying to mark outbound calls as anonymous and we set it to prohib, but calls still show outbound callerid. We are SIP from * to the Cisco gateway and then PRI outbound. If we strip the callerid num, then the first number on the PRI gets added as teh callerid, so we can't do that. We need to make

* --> SIP --> CISCO --> PRI --> PSTN The PSTN sees no callerid. *---> PRI[zaptel]--> PSTN Callerid is there... which makes me think it's the cisco, not the PRI/PSTN/telco. CISCO PRI--> * PRI [zaptel] Callerid IS there... which makes me shake my head in disbelief, because * can see clid from the cisco pri, but pstn doesn't... but when * sends info on that pri,

Hi guys. We have currently Asterisk CVS-v1-0-08/15/05-15:53:48 connected in SIP with a Cisco AS5300 (IOS 12.3). One PRI is connected to the Cisco gateway. The problem we have is that on incoming PSTN calls to the AS5300, relayed in SIP to Asterisk, the callerID name is not being transmitted. We received the callerID number but no name. I know we are receiving the name from the PRI in

I've got a very strange problem and I can't figure it out. I have a Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I see callerID name, but it is not getting to * via SIP. I am running * 1.4.2 and the latest Cisco IOS for my router. Here is what is happening: A call comes into the gateway. It sends a SIP INVITE to * with "pending" as the callerID

Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined

This depends on what you mean by ?not involving the service provider.? If you are literally forwarding calls that come in on the PRI back out on the PRI, the most efficient way is with Two B-Channel Transfer (TBCT). Check it out in the wiki. You need to make sure your carrier supports the feature. When you want to do a ?transfer,? you have an incoming call alerting or answered, you

This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message

My, how embarrassing. I of course meant that as a personal message to Don. But if anyone else knows the answer, I'm interested! lol Cheers, j On 03/18/2015 10:02 AM, Jeff LaCoursiere wrote: > > Hey Don, > > How are you? I may be heading your way in the next month or so. Have > to meet with a guy in Eden Prairie, and stop off at my > brother/sisterm-in-law's as

Does anybody have any experience using a Patton SmartNode as a SIP/Telco gateway with Asterisk? They seem really inexpensive and appear to support all of the necessary features, but I don't have any experience with their products, so I don't know if they are any good. We are currently using a Cisco 2600 w/ PRI card and it works fine, but I was looking for someone else as a possible

Greetings all.. We have an inbound/outbound PRI installed and terminated on a T400P ? Digium Quad T1 card. We?re seeing an odd problem when sending $CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN over the PRI. The $CALLERIDNUM is not being sent out along with the call. It?s sending the phone number of the PRI itself, rather than the $CALLERIDNUM information. Yes, we can

Hey Don, How are you? I may be heading your way in the next month or so. Have to meet with a guy in Eden Prairie, and stop off at my brother/sisterm-in-law's as well. Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? Lets say I take a call that was dialled locally (caller believes this is "free"), and I do a

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Cisco PRI card in a 7200 running lastest IOS --> SIP --> * Problem #1: SIP authenticates it under "general" and gives it the context from [general] not from [cisco.ip.address.here]. Here's the relevant party of sip.conf: [209.104.XX.YY] context=dids type=friend host=209.104.XX.YY ; IP address of Cisco gateway dtmfmode=rfc2833 disallow=all allow=ulaw [general] ...

I've got an Asterisk to H323 bridge working... but I'm having a few problems.. I got everything working by setting up with the Asterisk box as a gateway in CCM. I've got two issues.. 1. If I call off net.. (Asterisk -> CCM -> Cisco 5300(I think) -> PRI) the calls will proceed.. connect, and I get about 4-5 seconds of RTP and * tells me the remote end terminated my call. I

Doe anyone have Caller ID name working on incoming calls with the following setup: cisco 2611XM recieving calls on a PRI. cisco talks SIP to the asterisk box. Phones (cisco 7940s) talk SIP to asterisk. I currently have verified the cisco router is receiving caller ID name from the ILEC, and it appears to be sending it to the asterisk. But, it looks like it comes in after the call, and *

Hi, Trying to set up a SIP environment with Asterisk - T1/PRI on Cisco 2811 - PSTN (Teltone TSP). SIP call looks working and I am able to make call from SIP phones to PSTN analog phones. But I have troubles with the Call from PSTN to the SIP phones. I guess the problem is the dial plan. I am using four digits extensions (2301, 2302,..., 2xxx) on the Asterisk for the SIP phones and the full

Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough.

hi, after several problems getting the right callerid on a E1-PRI there is (so far) only one problem left: when receiving calls over the telephone network from another E1-PRI that has a "Caller ID no screen" capability (e.g. a bank and a customer of us), asterisk does not get the callerid that is set up by the calling PBX, but the callerid of the trunk of the calling PRI. no matter

XO fixed my caller ID name. I am using FreePBX and I can include a "wait" to my custom extensions. Is there a way to add a wait to the whole PRI? I assume that if I set immediate to yes, I can then have a "s" extension do the wait, but how would it get from the "s" to the DID extension? (also, I would rather not answer every call) Is there a "magic" spot