similar to: Cisco 7940 problems

Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). The problem I'm having is that when I connect to voicemail the DTMF key presses dont seem to work

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

I have noticed a problem with the Cisco 7940/7960 phones where if you put your voice-mail box on hold using soft keys and come back you can no longer navigate. I am curious if anyone else can duplicate this problem. Happens reliably for me with the 7940 phones. I use rfc2833 for DTMF. I would think it was a Cisco bug, but for the fact that this did not happen with older version of

Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823

> -----Original Message----- > From: Travis Johnson [mailto:tlj@ida.net] > Sent: Monday, September 08, 2003 1:05 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940/7960 ethernet ports > [...] > We are having a problem with Cisco 7940 and 7960 phones when > the PC is plugged into the 2nd ethernet port on the phone. It > will drop the

I'm having problems with the voicemail prompts. The beginning of each prompt is garbled. Prompts "edian mail" and "assword" can be recognized, but when it starts on the short phrases it is completely garbled. It is almost like the subsequent phrase starts before the current phrase finishes. I have isolated the LAN to the * box and the phones, so LAN congestion is

Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in

Has anyone had any major issues with the Cisco 7940 and or 7960 phones? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030818/2c156d1f/attachment.htm

can anyone give me a reference to the retrieval of voicemail from the Asterisk PBX using a cisco 7940 phine running sip image. i have configured a single voicemail box using the script, the corresponding entry in voicemail.conf and configured the extension to use the voicemail box . i can see from the asterisk console the message being passed to the voice mailbox, and correspondingly the sip

Hello all, this may be slightly offtopic :-) I have some Cisco 7940 phones with SIP firmware, connected to an Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g (HorstBox Pro with custom extensions.conf). On some of the phones, two lines are configured, one for business, one for private calls. When forwarding a line to another destination (e.g. to voicemail), we can't use the phone's own

Hi Siggi/Jan, >If so, there's still a load version conflict (although I've >never seen a >7960 or 7940 care about the version communicated through SCCP): > >On the phone, press "Settings", then 4 for load information. >watch out for the "App-Load-ID". On my 7940, this is >"P00305000300". Yours >is most likely a smaller number... >

Hi all, I'm having a strange problem with a bunch of cisco 7940G with SIP firmware. The problem arises when transferring a call coming in from a SIP account to another phone. The call connects, but for the first 10 seconds there is a situation with one-way audio, then it turns into a fully working call. I've googled extensively, but couldn't find much about this situation. The

Hi, I have encountered a DTMF issue. My scenario: Access carrier-----sip----> Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind

Hi list, I'm hoping that I'm being stupid, and someone can tell me what's going on, but for the life of me I can't figure it out. (it's been a long day, and I'm now in the last 3 weeks of organising my wedding, so I hope this makes sense ;) ) When at my desk, accessing (for example) my voicemail, the dtmf tones are passed perfectly, I can enter password, change

We're having an issue with connecting a Cisco ITS installation to * such that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or to any of the interfaces behind *. On the Cisco Side: dial-peer voice 8 voip destination-pattern 9999$ session protocol sipv2 session target ipv4:172.16.1.249 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad We have also

Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001>

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show

I have quite successfully set up the Services button to work on the 7940 running SIP. I have a metric-imperial converter, a foreign exchange rate calculator, a calendar etc available to users. The XML is really fussy though. Simon -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Rich Adamson Sent: Thursday, 1

We've just started testing with Asterisk (CVS HEAD) and a pair of Cisco 7940G phones running the SIP 6.3 firmware. One issue that we've run into is the ability to have multiple calls ring to the phone. Our scenario is that the user is using an extension and another call comes in for that extension. We'd like to have that second call ring the second line -- the same extension is

Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which