similar to: Voice only works from one way.

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

All, I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is.... One pri terminating into a Cisco 2431 router Sip messages from the Cisco get sent to a asterisk server linksys ata's a each remote end. I can receive the calling name if the call originates

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

If you have a PRI terminated in a Cisco router talking SIP to * and would be willing to share your Cisco config, please respond. Also, I would be interested in knowing what version of IOS you are using. We are using an NM-HDV in a 3640. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I'm experimenting (using for the first time) with using a cisco3620 to connect to the PSTN via a channelised E1 interface, with * handling all of the SIP calls. If anyone has any installation tips / help / documentation I would be most appreciative :) However, my first question is this: when I am in the setup, I see the following: Current interface summary Controller Timeslots

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

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

Hi, I am experiencing one-way audio from: SIP Device ----> Asterisk -----> Cisco 7200 The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass audio from SIP Device to Asterisk through the Cisco 7200 to the other end, but the Cisco 7200 does not return any audio back to the SIP Device or Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no

Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

Dear All It is a little hard to give a good small example of my question,so I will show the full data on the bottom and the attachment.Maybe some one could tell me an appropriate way to show it.I'm sorry for the inconvenience. Q:How to generate a 53*53 diagonal matrix by my data Some problems confused me are that: 1.Since it is a diagonal matrix,I have tried to transform col1 and col2 to

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

Im sure this has been thrown around this list 1,000 times, and Im sure its been around the net too.. But I have done everything, and cannot seem to get inward calls to be processed on my asterisk box.. First, Let me tell you what works: 1) Softphones (ZoIPer using IAX2 Protocol) Can make calls behind a Natted Firewall to the FXS Port, and it rings, and calls work full duplex. 2) Soyo IP Phone

Hi I have asterisk ip-pbx on my network, with some grandstream ip phone and i have cisco gateway that is connetced to VOIP service providers . Cisco is 3700 series and is using H323 . i have compiled H323 on asterisk . now i want to make a call from ip phone that is registerd to asterisk , and route call to VOIP provider so this call should goes to teh cisco from asterisk and then

Hello, We have a DID partner sending traffic to Asterisk via SIP, but we are not hearing ringtones. When we call the same extension via SIP, we can hear that's it"s ringing (virtually).. Is is something related with call-progress not recognized by DID provider ? Thanks, ________________________ a b d o u l aba at gcomnetworks.com SIP: (131) 229-1002 at sip.freeipcall.com

Hi i am search a sample config (for asterisk and for cisco) for connect a cisco 1760 with a FXO card to my asterisk. Thanks for your help Jerome

Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound & get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered

Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error

I have cisco phones running SCCP, and a cisco 2600 with FXO I'm using for PSTN access. I can dial out, but inbound calls are not ringing a phone. Please see my config In the 2600 I'm PLAR'ing the line and I have a SIP Dial-Peer for 4001 voice-port 1/1/0 output attenuation 0 echo-cancel coverage 32 no comfort-noise timing hookflash-out 50 connection plar opx 4001