similar to: Cisco 7961G + 7914 Expansion Module

All, I have run into some CallerID issues. It seems to have happened as a result of just moving my config from 1.2.12 to 1.4.4 (although I am not sure of this). Therefore I am sure its just a misconfiguration somewhere, I just don't know where. I have throughout the office either Cisco 7961G or Polycomm Soundpoint SIP 430 IP phones. The problem is with CallerID showing up in some

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 8f68421-22821e1e at localhost for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call 8f68421-22821e1e at localhost - no reply to our critical packet. any Ideas? Jason

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my

All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality,

All, I am having a lot of trouble with the Cisco 7961G phones. I have managed to get them up and running with Asterisk to the point where I can get incoming calls and make outgoing calls. The problem is when I make outgoing calls or extension to extension calls, the calls die after 20 seconds. I have google'd around and came up with little that is of help. The firmware version I am

Hi, Has anybody been successful in running the 7914 expansion unit for the Cisco 7960G IP phone? For anybody unaware of what the expansion unit does, it provides 14 additional buttons, with an LCD display. The idea, is that with an expansion unit (a 7960 can take upto 2 of these units), a user can either assign more speed-dial's, or can monitor line status/account status. So, you can

Hello, I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front of the phone and also to hopefully resolve some issues with the phones not registering after a long period. Once we upgraded the phones now display "Error Verifying Config Info" in the Status messages and will not process the

Hi, Does anyone have any experience with a Cisco 7961 + 7914 Operator Console setup and speeddials/BLF on Asterisk 1.4? Would appreciate feedback if this works reliably. I have a 7961 on skinny registering on an 1.4.19 box with chan_sccp and speeddials work fine so that part seems ok. I have no experience with the BLF part. >From googling around it seems that BLF on the 7914 only works with

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

Hello all, I'm in final testing stages and preparing training for a new Asterisk rollout. I'm replacing a Cisco Call Manager system, and re-flashing the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call, I use the "Confrn" softkey to invite other participants. I can add one other participant endpoint into the conference, but no more. I know I can (and

Hi All, We've been running Asterisk for years in our offices but just recently replaced an Asterisk Appliance* in our smaller office with an actual server, upgraded the server in hardware in our HQ location and upgrading both ends to 11.5.0 with Gareth's patch for Cisco phones. 99.99% of our endpoints are Cisco 7961Gs. Each office is more-or-less standalone for ease of management and

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

I found a few mentions of the 7914 being used with Asterisk, these all covered SCCP/skinny though. Does anyone know if the 7914 can even be used with SIP? If so, any pointers? Is it a services thing? Anyone get the operator (line/extension status) to work with it. Thanks for the help, Cisco doesn't even mention ANYTHING about SIP + the 7914. Matt

I am looking into the possibility of buying a Cisco 7960 with a 7914 expansion module. I know a lot of people are using the 7960, but I haven't read much about the 7914 and I was wondering if anybody has used this module with Asterisk? -- Thank you for your time __________________________________ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com

I'm still having problems getting this to work. I cannot get anything to display on my 7914 other than blank lines. I have SIP/5920-5930 in [main] that I'd like to add to the 7914 and indicate hook status. The 7960 is registering okay as SCCP/5000. What exactly should my sccp.conf file look like? When I make changes to this, how do I enact them? Do I reload Asterisk and reboot the phone

I was wondering if anyone had any experience getting a 7960+7914 working with any of the chan_sccp modules. I've got a 7960G with 6.0(5.0) and a factory fresh 7960G with 3.1(MF.G2). I've got 2 7914s fresh out of the box brand new. I hook them up and all I get is red lights on all of the buttons. When I go into the phone to see what version firmware they have it says "Link State:

Hello list, Has anyone on list had experience with getting the 7914 module working with Asterisk, probably using the chan-sccp drivers? I have 7912 and 7940 phones working OK, I need something for reception, and I was thinking a 7960 with two of these modules would handle it, but need to know that someone has got it working somewhere (seeing lines that are free, busy etc) Mikel

I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after